Pulse-code modulation (PCM) is a digital representation of an analog signal where the

magnitude of the signal is sampled regularly at uniform intervals, then quantized to a

series of symbols in a numeric (usually binary) code. PCM has been used in digital
telephone systems and 1980s-era electronic musical keyboards. It is also the standard
form for digital audio in computers and the compact disc "red book" format. It is also
standard in digital video, for example, using ITU-R BT.601. Uncompressed PCM is not
typically used for video in standard definition consumer applications such as DVD or
DVR because the bit rate required is far too high.

Modulation

Sampling and quantization of a signal (red) for 4-bit PCM

In the diagram, a sine wave (red curve) is sampled and quantized for PCM. The sine
wave is sampled at regular intervals, shown as ticks on the x-axis. For each sample, one
of the available values (ticks on the y-axis) is chosen by some algorithm (in this case, the
floor function is used). This produces a fully discrete representation of the input signal
(shaded area) that can be easily encoded as digital data for storage or manipulation. For
the sine wave example at right, we can verify that the quantized values at the sampling
moments are 7, 9, 11, 12, 13, 14, 14, 15, 15, 15, 14, etc. Encoding these values as binary
numbers would result in the following set of nibbles: 0111, 1001, 1011, 1100, 1101,
1110, 1110, 1111, 1111, 1111, 1110, etc. These digital values could then be further
processed or analyzed by a purpose-specific digital signal processor or general purpose
CPU. Several Pulse Code Modulation streams could also be multiplexed into a larger
aggregate data stream, generally for transmission of multiple streams over a single
physical link. This technique is called time-division multiplexing, or TDM, and is widely
used, notably in the modern public telephone system.

There are many ways to implement a real device that performs this task. In real systems,
such a device is commonly implemented on a single integrated circuit that lacks only the
clock necessary for sampling, and is generally referred to as an ADC (Analog-to-Digital
converter). These devices will produce on their output a binary representation of the input
whenever they are triggered by a clock signal, which would then be read by a processor
of some sort.

Demodulation
To produce output from the sampled data, the procedure of modulation is applied in
reverse. After each sampling period has passed, the next value is read and a signal is
shifted to the new value. As a result of these transitions, the signal will have a significant
amount of high-frequency energy. To smooth out the signal and remove these undesirable
aliasing frequencies, the signal would be passed through analog filters that suppress
energy outside the expected frequency range (that is, greater than the Nyquist frequency
fs / 2). Some systems use digital filtering to remove some of the aliasing, converting the
signal from digital to analog at a higher sample rate such that the analog filter required
for anti-aliasing is much simpler. In some systems, no explicit filtering is done at all; as
it's impossible for any system to reproduce a signal with infinite bandwidth, inherent
losses in the system compensate for the artifacts — or the system simply does not require
much precision. The sampling theorem suggests that practical PCM devices, provided a
sampling frequency that is sufficiently greater than that of the input signal, can operate
without introducing significant distortions within their designed frequency bands.

The electronics involved in producing an accurate analog signal from the discrete data are
similar to those used for generating the digital signal. These devices are DACs (digital-to-
analog converters), and operate similarly to ADCs. They produce on their output a
voltage or current (depending on type) that represents the value presented on their inputs.
This output would then generally be filtered and amplified for use.

Limitations
There are two sources of impairment implicit in any PCM system:

• Choosing a discrete value near the analog signal for each sample (quantization
error)

The quantization error swings between to . In the ideal case (with a fully linear ADC) it is
equally distributed over this interval thus with follows equals zero while the equals to

• Between samples no measurement of the signal is made; due to the sampling

theorem this results in any frequency above or equal to (fs being the sampling
frequency) being distorted or lost completely (aliasing error). This is also called
the Nyquist frequency.

As samples are dependent on time, an accurate clock is required for accurate

reproduction. If either the encoding or decoding clock is not stable, its frequency drift
will directly affect the output quality of the device. A slight difference between the
encoding and decoding clock frequencies is not generally a major concern; a small
constant error is not noticeable. Clock error does become a major issue if the clock is not
stable, however. A drifting clock, even with a relatively small error, will cause very
obvious distortions in audio and video signals, for example.

Digitization as part of the PC process

In conventional PCM, the analog signal may be processed (e.g. by amplitude
compression) before being digitized. Once the signal is digitized, the PCM signal is
usually subjected to further processing (e.g. digital data compression).
Some forms of PCM combine signal processing with coding. Older versions of these
systems applied the processing in the analog domain as part of the A/D process; newer
implementations do so in the digital domain. These simple techniques have been largely
rendered obsolete by modern transform-based audio compression techniques.

• DPCM encodes the PCM values as differences between the current and the
predicted value. An algorithm predicts the next sample based on the previous
samples, and the encoder stores only the difference between this prediction and
the actual value. If the prediction is reasonable, fewer bits can be used to represent
the same information. For audio, this type of encoding reduces the number of bits
required per sample by about 25% compared to PCM.
• Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the
quantization step, to allow further reduction of the required bandwidth for a given
signal-to-noise ratio.
• Delta modulation, another variant, uses one bit per sample.

In telephony, a standard audio signal for a single phone call is encoded as 8000 analog
samples per second, of 8 bits each, giving a 64 kbit/s digital signal known as DS0. The
default signal compression encoding on a DS0 is either μ-law (mu-law) PCM (North
America and Japan) or A-law PCM (Europe and most of the rest of the world). These are
logarithmic compression systems where a 12 or 13-bit linear PCM sample number is
mapped into an 8-bit value. This system is described by international standard G.711. An
alternative proposal for a floating point representation, with 5-bit mantissa and 3-bit
radix, was abandoned.

Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes
sense to compress the voice signal even further. An ADPCM algorithm is used to map a
series of 8-bit µ-law or A-law PCM samples into a series of 4-bit ADPCM samples. In
this way, the capacity of the line is doubled. The technique is detailed in the G.726
standard.

Later it was found that even further compression was possible and additional standards
were published. Some of these international standards describe systems and ideas which
are covered by privately owned patents and thus use of these standards requires payments
to the patent holders.

Encoding for transmission

Pulse-code modulation can be either return-to-zero (RZ) or non-return-to-zero (NRZ).
For a NRZ system to be synchronized using in-band information, there must not be long
sequences of identical symbols, such as ones or zeroes. For binary PCM systems, the
density of 1-symbols is called ones-density.

Ones-density is often controlled using precoding techniques such as Run Length Limited
encoding, where the PCM code is expanded into a slightly longer code with a guaranteed
bound on ones-density before modulation into the channel. In other cases, extra framing
bits are added into the stream which guarantee at least occasional symbol transitions.

Another technique used to control ones-density is the use of a scrambler polynomial on

the raw data which will tend to turn the raw data stream into a stream that looks pseudo-
random, but where the raw stream can be recovered exactly by reversing the effect of the
polynomial. In this case, long runs of zeroes or ones are still possible on the output, but
are considered unlikely enough to be within normal engineering tolerance.

In other cases, the long term DC value of the modulated signal is important, as building
up a DC offset will tend to bias detector circuits out of their operating range. In this case
special measures are taken to keep a count of the cumulative DC offset, and to modify the
codes if necessary to make the DC offset always tend back to zero.

Many of these codes are bipolar codes, where the pulses can be positive, negative or
absent. In the typical alternate mark inversion code, non-zero pulses alternate between
being positive and negative. These rules may be violated to generate special symbols
used for framing or other special purposes.

History
In the history of electrical communications, the earliest reason for sampling a signal was
to interlace samples from different telegraphy sources, and convey them over a single
telegraph cable. Telegraph time-division multiplexing (TDM) was conveyed as early as
1853, by the American inventor M.B. Farmer. The electrical engineer W.M. Miner, in
1903, used an electro-mechanical commutator for time-division multiplex of multiple
telegraph signals, and also applied this technology to telephony. He obtained intelligible
speech from channels sampled at a rate above 3500–4300 Hz: below this was
unsatisfactory. This was TDM, but pulse-amplitude modulation (PAM) rather than PCM.

Paul M. Rainey of Western Electric in 1926 patented a facsimile machine using an

optical mechanical analog to digital converter. The machine did not go into production.
British engineer Alec Reeves, unaware of previous work, conceived the use of PCM for
voice communication in 1937 while working for International Telephone and Telegraph
in France. He described the theory and advantages, but no practical use resulted. Reeves
filed for a French patent in 1938, and his U.S. patent was granted in 1943.

The first transmission of speech by digital techniques was the SIGSALY vocoder
encryption equipment used for high-level Allied communications during World War II
from 1943. In 1943, the Bell Labs researchers who designed the SIGSALY system,
became aware of the use of PCM binary coding as already proposed by Alec Reeves. In
1949 for the Canadian Navy's DATAR system, Ferranti Canada built a working PCM
radio system that was able to transmit digitized radar data over long distances.

PCM in the 1950s used a cathode-ray coding tube with a grid having encoding
perforations. As in an oscilloscope, the beam was swept horizontally at the sample rate
while the vertical deflection was controlled by the input analog signal, causing the beam
to pass through higher or lower portions of the perforated grid. The grid interrupted the
beam, producing current variations in binary code. Rather than natural binary, the grid
was perforated to produce Gray code lest a sweep along a transition zone produce
glitches.

Nomenclature
The word pulse in the term Pulse-Code Modulation refers to the "pulses" to be found in
the transmission line. This perhaps is a natural consequence of this technique having
evolved alongside two analog methods, pulse width modulation and pulse position
modulation, in which the information to be encoded is in fact represented by discrete
signal pulses of varying width or position, respectively. In this respect, PCM bears little
resemblance to these other forms of signal encoding, except that all can be used in time
division multiplexing, and the binary numbers of the PCM codes are represented as
electrical pulses. The device that performs the coding and decoding function in a
telephone circuit is called a code
Pulse-position modulation
Pulse-position modulation is a form of signal modulation in which M message bits
are encoded by transmitting a single pulse in one of 2M possible time-shifts. This is
repeated every T seconds, such that the transmitted bit rate is M/T bits per second. It is
primarily useful for optical communications systems, where there tends to be little or no
multipath interference.

Synchronization
One of the key difficulties of implementing this technique is that the receiver must be
properly synchronized to align the local clock with the beginning of each symbol.
Therefore, it is often implemented differentially as Differential Pulse-position
modulation, where by each pulse position is encoded relative to the previous , such that
the receiver must only measure the difference in the arrival time of successive pulses. It is
possible to limit the propagation of errors to adjacent symbols, so that an error in
measuring the differential delay of one pulse will affect only two symbols, instead of
effecting all successive measurements.

Sensitivity to Multipath Interference

Aside from the issues regarding receiver synchronization, the key disadvantage of PPM is
that it is inherently sensitive to multipath interference that arises in channels with
frequency-selective fading, whereby the receiver's signal contains one or more echoes of
each transmitted pulse. Since the information is encoded in the time of arrival (either
differentially, or relative to a common clock), the presence of one or more echoes can
make it extremely difficult, if not impossible, to accurately determine the correct pulse
position corresponding to the transmitted pulse.

Non-coherent Detection
One of the principal advantages of Pulse Position Modulation is that it is an M-ary
modulation technique that can be implemented non-coherently, such that the receiver
does not need to use a Phase-locked loop (PLL) to track the phase of the carrier. This
makes it a suitable candidate for optical communications systems, where coherent phase
modulation and detection are difficult and extremely expensive. The only other common
M-ary non-coherent modulation technique is M-ary Frequency Shift Keying, which is the
frequency-domain dual to PPM.

PPM vs. M-FSK

PPM and M-FSK systems with the same bandwidth, average power, and transmission
rate of M/T bits per second have identical performance in an AWGN (Additive White
Gaussian Noise) channel. However, their performance differs greatly when comparing
frequency-selective and frequency-flat fading channels. Whereas frequency-selective
fading produces echoes that are highly disruptive for any of the M time-shifts used to
encode PPM data, it selectively disrupts only some of the M possible frequency-shifts
used to encode data for M-FSK. Conversely, frequency-flat fading is more disruptive for
M-FSK than PPM, as all M of the possible frequency-shifts are impaired by fading, while
the short duration of the PPM pulse means that only a few of the M time-shifts are
heavily impaired by fading.

Optical communications systems (even wireless ones) tend to have weak multipath
distortions, and PPM is a viable modulation scheme in many such applications.

Applications for RF Communications

PPM Implementation
This figure illustrates how PPM is used to control servos in RC applications.
N.B. Unfortunately this image does not display PPM, but a PWM signal

Narrowband RF (Radio Frequency) channels with low power and long wavelengths (i.e.,
low frequency) are affected primarily by flat fading, and PPM is better suited than M-
FSK to be used in these scenarios. One common application with these channel
characteristics, first used in the early 1960s, is the radio control of model aircraft, boats
and cars. PPM is employed in these systems, with the position of each pulse representing
the angular position of an analogue control on the transmitter, or possible states of a
binary switch. The number of pulses per frame gives the number of controllable channels
available. The advantage of using PPM for this type of application is that the electronics
required to decode the signal are extremely simple, which leads to small, light-weight
receiver/decoder units. (Model aircraft require parts that are as lightweight as possible).
Servos made for model radio control include some of the electronics required to convert
the pulse to the motor position - the receiver is merely required to demultiplex the
separate channels and feed the pulses to each servo.

More sophisticated R/C systems are now often based on pulse-code modulation, which is
more complex but offers greater flexibility and reliability.

Overview
In wireless telecommunications, multipath is the propagation phenomenon that results in
radio signals' reaching the receiving antenna by two or more paths. Causes of multipath
include atmospheric ducting, ionospheric reflection and refraction, and reflection from
water bodies and terrestrial objects such as mountains and buildings.

The effects of multipath include constructive and destructive interference, and phase
shifting of the signal. This causes Rayleigh fading, named after Lord Rayleigh. The
standard statistical model of this gives a distribution known as the Rayleigh distribution.

Rayleigh fading with a strong line of sight content is said to have a Rician distribution, or
to be Rician fading.

In facsimile and television transmission, multipath causes jitter and ghosting, seen as a
faded duplicate image to the right of the main image. Ghosts occur when transmissions
bounce off a mountain or other large object, while also arriving at the antenna by a
shorter, direct route, with the receiver picking up two signals separated by a delay.

Radar multipath echoes from an actual target cause ghosts to appear.

In radar processing, multipath causes ghost targets to appear, deceiving the radar
receiver. These ghosts are particularly bothersome since they move and behave like the
normal targets (which they echo), and so the receiver has difficulty in isolating the
correct target echo. These problems can be overcome by incorporating a ground map of
the radar's surroundings and eliminating all echoes which appear to originate below
ground or above a certain height.

In digital radio communications (such as GSM) multipath can cause errors and affect the
quality of communications. The errors are due to Intersymbol interference (ISI).
Equalisers are often used to correct the ISI. Alternatively, techniques such as orthogonal
frequency division modulation and Rake receivers may be used.

In a Global Positioning System receiver, multipath signals can cause a stationary

receiver's output to indicate as if it were randomly jumping about or creeping. When the
unit is moving the jumping or creeping is hidden, but it still degrades the displayed
accuracy.

Multipath propagation in wired media

Multipath propagation may also happen in wired media, specially in cases in which
impedance mismatches cause signal reflections. A well-known example is Power line
communication.

High-speed Power line communication systems usually employ multi-carrier modulations

(such as OFDM or Wavelet OFDM) to avoid the Intersymbol interference that Multipath
propagation would cause.

The ITU-T G.hn standard provides a way to create a high-speed (up to 1 Gigabit/s) Local
area network using existing home wiring (power lines, phone lines and coaxial cables).
G.hn uses OFDM with a Cyclic prefix to avoid ISI. Because Multipath propagation
behaves differently in each kind of wire, G.hn uses different OFDM parameters (OFDM
symbol duration, Guard Interval duration) for each media.

Mathematical modeling

Mathematical model of the multipath impulse response.

The mathematical model of the multipath can be presented using the method of the
impulse response used for studying linear systems.

Suppose to transmit a single, ideal Dirac pulse of electromagnetic power at time 0, i.e.

x(t) = δ(t)
At the receiver, due to the presence of the multiple electromagnetic paths, more than one
pulse will be received (we suppose here that the channel has infinite bandwidth, thus the
pulse shape is not modified at all), and each one of them will arrive at different times. In
fact, since the electromagnetic signals travel at the speed of light, and since every path
has a geometrical length possibly different from that of the other ones, there are different
air travelling times (consider that, in free space, the light takes 3μs to cross a 1 km span).
Thus, the received signal will be expressed by

where N is the number of received impulses (equivalent to the number of

electromagnetic paths, and possibly very large), τn is the time delay of the generic nth
impulse, and represent the complex amplitude (i.e., magnitude and phase) of the generic
received pulse. As a consequence, y(t) also represents the impulse response function
h(t) of the equivalent multipath model.
More in general, in presence of time variation of the geometrical reflection conditions,
this impulse response is time varying, and as such we have

τn = τn(t)
ρn = ρn(t)
φn = φn(t)
Very often, just one parameter is used to denote the severity of multipath conditions: it is
called the multipath time, TM, and it is defined as the time delay existing between the
first and the last received impulses

TM = τN − 1 − τ0

Mathematical model of the multipath channel transfer function.

In practical conditions and measurement, the multipath time is computed by considering

as last impulse the first one which allows to receive a determined amount of the total
transmitted power (scaled by the atmospheric and propagation losses), e.g. 99%.

Keeping our aim at linear, time invariant systems, we can also characterize the multipath
phenomenon by the channel transfer function H(f), which is defined as the continuous
time Fourier transform of the impulse response h(t)

where the last right-hand term of the previous equation is easily obtained by
remembering that the Fourier transform of a Dirac pulse is a complex exponential
function, an eigenfunction of every linear system.

The obtained channel transfer characteristic has a typical appearance of a sequence of

peaks and valleys (also called notches); it can be shown that, on average, the distance (in
Hz) between two consecutive valleys (or two consecutive peaks), is roughly inversely
proportional to the multipath time. The so-called coherence bandwidth is thus defined
as

For example, with a multipath time of 3μs (corresponding to a 1 km of added on-air

travel for the last received impulse), there is a coherence bandwidth of about 330 kHz
The Nyquist–Shannon sampling theorem is a fundamental result in the field of
information theory, in particular telecommunications and signal processing. Sampling is
the process of converting a signal (for example, a function of continuous time or space)
into a numeric sequence (a function of discrete time or space). The theorem states:[1]

If a function x(t) contains no frequencies higher than B hertz, it is completely determined by

giving its ordinates at a series of points spaced 1/(2B) seconds apart.

In essence the theorem shows that an analog signal that has been sampled can be
perfectly reconstructed from the samples if the sampling rate exceeds 2B samples per
second, where B is the highest frequency in the original signal. If a signal contains a
component at exactly B hertz, then samples spaced at exactly 1/(2B) seconds do not
completely determine the signal, Shannon's statement notwithstanding.

More recent statements of the theorem are sometimes careful to exclude the equality
condition; that is, the condition is if x(t) contains no frequencies higher than or equal to
B; this condition is equivalent to Shannon's except when the function includes a steady
sinusoidal component at exactly frequency B.

The assumptions necessary to prove the theorem form a mathematical model that is only
an idealization of any real-world situation. The conclusion that perfect reconstruction is
possible is mathematically correct for the model but only an approximation for actual
signals and actual sampling techniques.

The theorem also leads to a formula for reconstruction of the original signal. The
constructive proof of the theorem leads to an understanding of the aliasing that can occur
when a sampling system does not satisfy the conditions of the theorem.

The Nyquist–Shannon sampling theorem is also known to be a sufficient condition.

The field of Compressed sensing provides a stricter sampling condition when the
underlying signal is known to be sparse. Compressed sensing specifically yields a sub-
Nyquist sampling criterion.

Introduction
A signal or function is bandlimited if it contains no energy at frequencies higher than
some bandlimit or bandwidth B. A signal that is bandlimited is constrained in how
rapidly it changes in time, and therefore how much detail it can convey in an interval of
time. The sampling theorem asserts that the uniformly spaced discrete samples are a
complete representation of the signal if this bandwidth is less than half the sampling rate.
To formalize these concepts, let represent a continuous-time signal and be the Fourier
transform of that signal:

The signal is bandlimited to a one-sided baseband bandwidth, B, if:

 for all

or, equivalently, supp(X)[2] [-B,B]. Then the sufficient condition for exact
reconstructability from samples at a uniform sampling rate (in samples per unit time) is:

or equivalently:

is called the Nyquist rate and is a property of the bandlimited signal, while is called the
Nyquist frequency and is a property of this sampling system.

The time interval between successive samples is referred to as the sampling interval:

and the samples of are denoted by:

(integers).

The sampling theorem leads to a procedure for reconstructing the original from the
samples and states sufficient conditions for such a reconstruction to be exact.

The sampling process

The theorem describes two processes in signal processing: a sampling process, in which a
continuous time signal is converted to a discrete time signal, and a reconstruction
process, in which the original continuous signal is recovered from the discrete time
signal.

The continuous signal varies over time (or space in a digitized image, or another
independent variable in some other application) and the sampling process is performed by
measuring the continuous signal's value every T units of time (or space), which is called
the sampling interval. In practice, for signals that are a function of time, the sampling
interval is typically quite small, on the order of milliseconds, microseconds, or less. This
results in a sequence of numbers, called samples, to represent the original signal. Each
sample value is associated with the instant in time when it was measured. The reciprocal
of the sampling interval (1/T) is the sampling frequency denoted fs, which is measured in
samples per unit of time. If T is expressed in seconds, then fs is expressed in Hz.

Reconstruction of the original signal is an interpolation process that mathematically

defines a continuous-time signal x(t) from the discrete samples x[n] and at times in
between the sample instants nT.

Fig.2: The normalized sinc function: sin(πx) / (πx) ... showing the central peak at x= 0,
and zero-crossings at the other integer values of x.
 • The procedure: Each sample value is multiplied by the sinc function scaled so
that the zero-crossings of the sinc function occur at the sampling instants and that
the sinc function's central point is shifted to the time of that sample, nT. All of
these shifted and scaled functions are then added together to recover the original
signal. The scaled and time-shifted sinc functions are continuous making the sum
of these also continuous, so the result of this operation is a continuous signal. This
procedure is represented by the Whittaker–Shannon interpolation formula.

• The condition: The signal obtained from this reconstruction process can have no
frequencies higher than one-half the sampling frequency. According to the
theorem, the reconstructed signal will match the original signal provided that the
original signal contains no frequencies at or above this limit. This condition is
called the Nyquist criterion, or sometimes the Raabe condition.

If the original signal contains a frequency component equal to one-half the sampling rate,
the condition is not satisfied. The resulting reconstructed signal may have a component at
that frequency, but the amplitude and phase of that component generally will not match
the original component.

This reconstruction or interpolation using sinc functions is not the only interpolation
scheme. Indeed, it is impossible in practice because it requires summing an infinite
number of terms. However, it is the interpolation method that in theory exactly
reconstructs any given bandlimited x(t) with any bandlimit B < 1/2T); any other method
that does so is formally equivalent to it.

Practical considerations
A few consequences can be drawn from the theorem:

• If the highest frequency B in the original signal is known, the theorem gives the
lower bound on the sampling frequency for which perfect reconstruction can be
assured. This lower bound to the sampling frequency, 2B, is called the Nyquist
rate.

• If instead the sampling frequency is known, the theorem gives us an upper bound
for frequency components, B<fs/2, of the signal to allow for perfect
reconstruction. This upper bound is the Nyquist frequency, denoted fN.

• Both of these cases imply that the signal to be sampled must be bandlimited; that
is, any component of this signal which has a frequency above a certain bound
should be zero, or at least sufficiently close to zero to allow us to neglect its
influence on the resulting reconstruction. In the first case, the condition of
bandlimitation of the sampled signal can be accomplished by assuming a model of
the signal which can be analysed in terms of the frequency components it
contains; for example, sounds that are made by a speaking human normally
contain very small frequency components at or above 10 kHz and it is then
 sufficient to sample such an audio signal with a sampling frequency of at least 20
kHz. For the second case, we have to assure that the sampled signal is bandlimited
such that frequency components at or above half of the sampling frequency can be
neglected. This is usually accomplished by means of a suitable low-pass filter; for
example, if it is desired to sample speech waveforms at 8 kHz, the signals should
first be lowpass filtered to below 4 kHz.

• In practice, neither of the two statements of the sampling theorem described

above can be completely satisfied, and neither can the reconstruction formula be
precisely implemented. The reconstruction process that involves scaled and
delayed sinc functions can be described as ideal. It cannot be realized in practice
since it implies that each sample contributes to the reconstructed signal at almost
all time points, requiring summing an infinite number of terms. Instead, some
type of approximation of the sinc functions, finite in length, has to be used. The
error that corresponds to the sinc-function approximation is referred to as
interpolation error. Practical digital-to-analog converters produce neither scaled
and delayed sinc functions nor ideal impulses (that if ideally low-pass filtered
would yield the original signal), but a sequence of scaled and delayed rectangular
pulses. This practical piecewise-constant output can be modeled as a zero-order
hold filter driven by the sequence of scaled and delayed dirac impulses referred to
in the mathematical basis section below. A shaping filter is sometimes used after
the DAC with zero-order hold to make a better overall approximation.

• Furthermore, in practice, a signal can never be perfectly bandlimited, since ideal

"brick-wall" filters cannot be realized. All practical filters can only attenuate
frequencies outside a certain range, not remove them entirely. In addition to this, a
"time-limited" signal can never be bandlimited. This means that even if an ideal
reconstruction could be made, the reconstructed signal would not be exactly the
original signal. The error that corresponds to the failure of bandlimitation is
referred to as aliasing.

• The sampling theorem does not say what happens when the conditions and
procedures are not exactly met, but its proof suggests an analytical framework in
which the non-ideality can be studied. A designer of a system that deals with
sampling and reconstruction processes needs a thorough understanding of the
signal to be sampled, in particular its frequency content, the sampling frequency,
how the signal is reconstructed in terms of interpolation, and the requirement for
the total reconstruction error, including aliasing and interpolation error. These
properties and parameters may need to be carefully tuned in order to obtain a
useful system.

Aliasing
Main article: Aliasing
The Poisson summation formula indicates that the samples of function x(t) are sufficient
to create a periodic extension of function X(f). The result is:

As depicted in Figures 3, 4, and 8, copies of X(f) are shifted by multiples of and

combined by addition.

Fig.3: Hypothetical spectrum of a properly sampled bandlimited signal (blue) and images
(green) that do not overlap. A "brick-wall" low-pass filter can remove the images and
leave the original spectrum, thus recovering the original signal from the samples.

If the sampling condition is not satisfied, adjacent copies overlap, and it is not possible in
general to discern an unambiguous X(f). Any frequency component above is
indistinguishable from a lower-frequency component, called an alias, associated with one
of the copies. The reconstruction technique described below produces the alias, rather
than the original component, in such cases.

Fig.4 Top: Hypothetical spectrum of an insufficiently sampled bandlimited signal (blue),

X(f), where the images (green) overlap. These overlapping edges or "tails" of the images
add, creating a spectrum unlike the original. Bottom: Hypothetical spectrum of a
marginally sufficiently sampled bandlimited signal (blue), XA(f), where the images
(green) narrowly do not overlap. But the overall sampled spectrum of XA(f) is identical to
the overall inadequately sampled spectrum of X(f) (top) because the sum of baseband and
images are the same in both cases. The discrete sampled signals xA[n] and x[n] are also
identical. It is not possible, just from examining the spectra (or the sampled signals), to
tell the two situations apart. If this were an audio signal, xA[n] and x[n] would sound the
same and the presumed "properly" sampled xA[n] would be the alias of x[n] since the
spectrum XA(f) masquerades as the spectrum X(f).

For a sinusoidal component of exactly half the sampling frequency, the component will in
general alias to another sinusoid of the same frequency, but with a different phase and
amplitude.

To prevent or reduce aliasing, two things can be done:

1. Increase the sampling rate, to above twice some or all of the frequencies that are
aliasing.
2. Introduce an anti-aliasing filter or make the anti-aliasing filter more stringent.

The anti-aliasing filter is to restrict the bandwidth of the signal to satisfy the condition for
proper sampling. Such a restriction works in theory, but is not precisely satisfiable in
reality, because realizable filters will always allow some leakage of high frequencies.
However, the leakage energy can be made small enough so that the aliasing effects are
negligible

From Wikipedia, the free encyclopedia

Jump to: navigation, search

Phase modulation (PM) is a form of modulation that represents information as

variations in the instantaneous phase of a carrier wave.

Unlike its more popular counterpart, frequency modulation (FM), PM is not very widely
used. This is because it tends to require more complex receiving hardware and there can
be ambiguity problems in determining whether, for example, the signal has changed
phase by +180° or -180°.

Theory
An example of phase modulation. The top diagram shows the modulating signal
superimposed on the carrier wave. The bottom diagram shows the resulting phase-
modulated signal.

Suppose that the signal to be sent (called the modulating or message signal) is m(t).

The carrier onto which the signal is to be modulated is

Then the modulated signal is

This shows how m(t) modulates the phase. Clearly, it can also be viewed as a change of
the frequency of the carrier signal. PM can thus be considered a special case of FM in
which the carrier frequency modulation is given by the time derivative of the phase
modulation.

The spectral behaviour of phase modulation is difficult to derive, but the mathematics
reveals that there are two regions of particular interest:

• For small amplitude signals, PM is similar to amplitude modulation (AM) and

exhibits its unfortunate doubling of baseband bandwidth and poor efficiency.
• For a single large sinusoidal signal, PM is similar to FM, and its bandwidth is
approximately

,
where fM = ωm / 2π and h is the modulation index defined below. This is also
known as Carson's Rule for PM.

Pulse-density modulation, or PDM, is a form of modulation used to represent an analog

signal in the digital domain. In a PDM signal, specific amplitude values are not encoded
into pulses as they would be in PCM. Instead it is the relative density of the pulses that
corresponds to the analog signal's amplitude. Pulse-width modulation (PWM) is the
special case of PDM where all the pulses corresponding to one sample are contiguous in
the digital signal.

Basics
In a pulse-density modulation bitstream a 1 corresponds to a pulse of positive polarity
(+A) and a 0 corresponds to a pulse of negative polarity (-A). Mathematically, this can be
represented as:

where x[n] is the bipolar bitstream (either -A or +A) and a[n] is the corresponding
binary bitstream (either 0 or 1).

A run consisting of all 1s would correspond to the maximum (positive) amplitude value,
all 0s would correspond to the minimum (negative) amplitude value, and alternating 1s
and 0s would correspond to a zero amplitude value. The continuous amplitude waveform
is recovered by low-pass filtering the bipolar PDM bitstream.

Analog-to-digital conversion
A PDM bitstream is encoded from an analog signal through the process of delta-sigma
modulation. This process uses a one bit quantizer that produces either a 1 or 0 depending
on the amplitude of the analog signal. A 1 or 0 corresponds to a signal that is all the way
up or all the way down, respectively. Because in the real world analog signals are rarely
all the way in one direction there is a quantization error, the difference between the 1 or 0
and the actual amplitude it represents. This error is fed back negatively in the ΔΣ process
loop. In this way every error successively influences every other quantization
measurement and its error. This has the effect of averaging out the quantization error.

Digital-to-analog conversion
The process of decoding a PDM signal into an analog one is simple: one only has to pass
the PDM signal through an analog low-pass filter. This works because the function of a
low-pass filter is essentially to average the signal. The average amplitude of pulses is
measured by the density of those pulses over time, thus a low pass filter is the only step
required in the decoding process.

Examples
A single period of the trigonometric sine function, sampled 100 times and represented as
a PDM bitstream, is:

01010110111101111111111111111111110111111011011010101001001000000100000
00000000000000001000010010101

An example of PDM of 100 samples of one period a sine wave. 1s represented by blue,
0s represented by white, overlaid with the sine wave.

Two periods of a higher frequency sine wave would appear as:

01011011111111111111011010100100000000000001000100110111011111111111110
11010100100000000000000100101 or

In pulse-density modulation, a high density of 1s occurs at the peaks of the sine wave,
while a low density of 1s occurs at the troughs of the sine wave.

Algorithm
Pulse-density modulation of a sine wave using this algorithm.

A digital model of pulse-density modulation can be obtained from a digital model of the
delta-sigma modulator. Consider a signal x[n] in the discrete time domain as the input to
a first-order delta-sigma modulator, with y[n] the output. In the discrete frequency
domain, the delta-sigma modulator's operation is represented by

Rearranging terms, we obtain

Here, E(z) is the frequency-domain quantization error of the delta-sigma modulator. The
factor 1 − z − 1 represents a high-pass filter, so it is clear that E(z) contributes less to the
output Y(z) at low frequencies, and more at high frequencies. This demonstrates the
noise shaping effect of the delta-sigma modulator: the quantization noise is "pushed" out
of the low frequencies up into the high-frequency range.

Using the inverse Z-transform, we may convert this into a difference equation relating the
input of the delta-sigma modulator to its output in the discrete time domain,

y[n] = x[n] + e[n] − e[n − 1]

There are two additional constraints to consider: first, at each step the output sample y[n]
is chosen so as to minimize the "running" quantization error e[n]. Second, y[n] is
represented as a single bit, meaning it can take on only two values. We choose
for convenience, allowing us to write
 e[n] = y[n] − x[n] + e[n − 1]

This, finally, gives a formula for the output sample y[n] in terms of the input sample
x[n]. The quantization error of each sample is fed back into the input for the following
sample.

The following pseudo-code implements this algorithm to convert a pulse-code

modulation signal into a PDM signal:

Library > Miscellaneous > Wikipedia

Sampled signal (discrete signal): discrete time, continuous values.

Quantized signal: continuous time, discrete values.

Digital signal (sampled, quantized): discrete time, discrete values.

In digital signal processing, quantization is the process of approximating ("mapping") a

continuous range of values (or a very large set of possible discrete values) by a relatively
small ("finite") set of ("values which can still take on continuous range") discrete
symbols or integer values. For example, rounding a real number in the interval [0,100] to
an integer

In other words, quantization can be described as a mapping that represents a finite

continuous interval I = [a,b] of the range of a continuous valued signal, with a single
number c, which is also on that interval. For example, rounding to the nearest integer
(rounding ½ up) replaces the interval [c − .5,c + .5) with the number c, for integer c.
After that quantization we produce a finite set of values which can be encoded by say
binary techniques.

In signal processing, quantization refers to approximating the output by one of a discrete

and finite set of values, while replacing the input by a discrete set is called discretization,
and is done by sampling: the resulting sampled signal is called a discrete signal (discrete
time), and need not be quantized (it can have continuous values). To produce a digital
signal (discrete time and discrete values), one both samples (discrete time) and quantizes
the resulting sample values (discrete values).

Contents
[show]

Applications
A common use of quantization is in the conversion of a discrete signal (a sampled
continuous signal) into a digital signal by quantizing. Both of these steps (sampling and
quantizing) are performed in analog-to-digital converters with the quantization level
specified in bits. A specific example would be compact disc (CD) audio which is sampled
at 44,100 Hz and quantized with 16 bits (2 bytes) which can be one of 65,536 (i.e. 216)
possible values per sample.

In electronics, adaptive quantization is a quantization process that varies the step size
based on the changes of the input signal, as a means of efficient compression. Two
approaches commonly used are forward adaptive quantization and backward adaptive
quantization.

Definition
In signal processing the quantization process is the necessary and natural follower of the
sampling operation. It is necessary because in practice the digital computer with is
general purpose CPU is used to implement DSP algorithms. And since computers can
only process finite word length (finite resolution/precision) quantities, any infinite
precision continuous valued signal should be quantized to fit a finite resolution, so that it
can be represented (stored) in CPU registers and memory.

We shall be aware of the fact that, it is not the continuous values of the analog function
that inhibits its binary encoding, rather it is the existence of infinitely many such values
due to the definition of continuity,(which therefore requires infinitely many bits to
represent). For example we can design a quantizer such that it represents a signal with a
single bit (just two levels) such that, one level is "pi=3,14..." (say encoded with a 1) and
the other level is "e=2.7183..." ( say encoded with a 0), as we can see, the quantized
values of the signal take on infinite precision, irrational numbers. But there are only two
levels. And we can represent the output of the quantizer with a binary symbol.
Concluding from this we can see that it is not the discreteness of the quantized values that
enable them to be encoded but the finiteness enabling the encoding with finite number of
bits.

In theory there is no relation between quantization values and binary code words used to
encode them (rather than a table that shows the corresponding mapping, just as
examplified above). However due to practical reasons we may tend to use code words
such that their binary mathematical values has a relation with the quantization levels that
is encoded. And this last option merges the first two paragrahs in such a way that, if we
wish to process the output of a quantizer within a DSP/CPU system (which is always the
case) then we can not allow the representation levels of the quantizers to take on arbitrary
values, but only a restricted range such that they can fit in computer registers.

A quantizer is identified with its number of levels M, the decision boundaries {di} and
the corresponding representation values {ri}.

The output of a quantizer has two important properties: 1) a Distortion resulting from the
approximation and 2) a Bit-Rate resulting from binary encoding of its levels. Therefore
the Quantizer design problem is a Rate-Distortion optimization type.

If we are only allowed to use fixed length code for the output level encoding (the
practical case) then the problem reduces into a distortion minimization one.

The design of a quantizer usually means the process to find the sets {di} and {ri} such
that a measure of optimality is satisfied (such as MMSEQ (Minimum Mean Squarred
Quantization Error))

Given the number of levels M, the optimal quantizer which minimizes the MSQE wrt the
given signal statistics is called the Max-Lloyd quantizer, which is a non-uniform type in
general.

The most common quantizer type is the uniform one. It is simple to design and
implement and for most cases it suffices to get satisfactory results. Indeed by the very
inherent nature of the design process, a given quantizer will only produce optimal results
for the assumed signal statistics. Since it is very difficult to correctly predict that in
advance, any static design will never produce actual optimal performance whenever the
input statistics deviates from that of the design assumption. The only solution is to use an
adaptive quantizer.

Mathematical description
Quantization is referred to as scalar quantization, since it operates on scalar (as opposed
to multi-dimensional vector) input data. In general, a scalar quantization operator can be
represented as
whereng an integer result that is sometimes referred to as the quantization
index,

• f(x) and g(i) are arbitrary real-valued functions.

The integer-valued quantization index i is the representation that is typically stored or

transmitted, and then the final interpretation is constructed using g(i) when the data is
later interpreted.

In computer audio and most other applications, a method known as uniform quantization
is the most common. There are two common variations of uniform quantization, called
mid-rise and mid-tread uniform quantizers.

If x is a real-valued number between -1 and 1, a mid-rise uniform quantization operator

that uses M bits of precision to represent each quantization index can be expressed as

In this case the f(x) and g(i) operators are just multiplying scale factors (one multiplier
being the inverse of the other) along with an offset in g(i) function to place the
representation value in the middle of the input region for each quantization index. The
value 2 − (M − 1) is often referred to as the quantization step size. Using this quantization law
and assuming that quantization noise is approximately uniformly distributed over the
quantization step size (an assumption typically accurate for rapidly varying x or high M)
and further assuming that the input signal x to be quantized is approximately uniformly
distributed over the entire interval from -1 to 1, the signal to noise ratio (SNR) of the
quantization can be computed via the 20 log rule as

From this equation, it is often said that the SNR is approximately 6 dB per bit.

For mid-tread uniform quantization, the offset of 0.5 would be added within the floor
function instead of outside of it.

Sometimes, mid-rise quantization is used without adding the offset of 0.5. This reduces
the signal to noise ratio by approximately 6.02 dB, but may be acceptable for the sake of
simplicity when the step size is small.
In digital telephony, two popular quantization schemes are the 'A-law' (dominant in
Europe) and 'μ-law' (dominant in North America and Japan). These schemes map discrete
analog values to an 8-bit scale that is nearly linear for small values and then increases
logarithmically as amplitude grows. Because the human ear's perception of loudness is
roughly logarithmic, this provides a higher signal to noise ratio over the range of audible
sound intensities for a given number of bits.

Quantization and data compression

Quantization plays a major part in lossy data compression. In many cases, quantization
can be viewed as the fundamental element that distinguishes lossy data compression from
lossless data compression, and the use of quantization is nearly always motivated by the
need to reduce the amount of data needed to represent a signal. In some compression
schemes, like MP3 or Vorbis, compression is also achieved by selectively discarding
some data, an action that can be analyzed as a quantization process (e.g., a vector
quantization process) or can be considered a different kind of lossy process.

One example of a lossy compression scheme that uses quantization is JPEG image
compression. During JPEG encoding, the data representing an image (typically 8-bits for
each of three color components per pixel) is processed using a discrete cosine transform
and is then quantized and entropy coded. By reducing the precision of the transformed
values using quantization, the number of bits needed to represent the image can be
reduced substantially. For example, images can often be represented with acceptable
quality using JPEG at less than 3 bits per pixel (as opposed to the typical 24 bits per pixel
needed prior to JPEG compression). Even the original representation using 24 bits per
pixel requires quantization for its PCM sampling structure.

In modern compression technology, the entropy of the output of a quantizer matters more
than the number of possible values of its output (the number of values being 2M in the
above example).

In order to determine how many bits are necessary to effect a given precision, algorithms
are used. Suppose, for example, that it is necessary to record six significant digits, that is
to say, millionths. The number of values that can be expressed by N bits is equal to two to
the Nth power. To express six decimal digits, the required number of bits is determined
by rounding (6 / log 2)—where log refers to the base ten, or common, logarithm—up to
the nearest integer. Since the logarithm of 2, base ten, is approximately 0.30102, the
required number of bits is then given by (6 / 0.30102), or 19.932, rounded up to the
nearest integer, viz., 20 bits.

This type of quantization—where a set of binary digits, e.g., an arithmetic register in a

CPU, are used to represent a quantity—is called Vernier quantization. It is also possible,
although rather less efficient, to rely upon equally spaced quantization levels. This is only
practical when a small range of values is expected to be captured: for example, a set of
eight possible values requires eight equally spaced quantization levels—which is not
unreasonable, although obviously less efficient than a mere trio of binary digits (bits)—
but a set of, say, sixty-four possible values, requiring sixty-four equally spaced
quantization levels, can be expressed using only six bits, which is obviously far more
efficient.

Relation to quantization in nature

At the most fundamental level, some physical quantities are quantized. This is a result of
quantum mechanics (see Quantization (physics)). Signals may be treated as continuous
for mathematical simplicity by considering the small quantizations as negligible.

In any practical application, this inherent quantization is irrelevant for two reasons. First,
it is overshadowed by signal noise, the intrusion of extraneous phenomena present in the
system upon the signal of interest. The second, which appears only in measurement
applications, is the inaccuracy of instruments. Thus, although all physical signals are
intrinsically quantized, the error introduced by modeling them as continuous is
vanishingly small.

, also used in other fields (such as scientific measurements, biological cell signaling),
defined as the ratio of a signal power to the noise power corrupting the signal.

In less technical terms, signal-to-noise ratio compares the level of a desired signal (such
as music) to the level of background noise. The higher the ratio, the less obtrusive the
background noise is.

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Technical sense
In engineering, signal-to-noise ratio is a term for the power ratio between a signal
(meaningful information) and the background noise:

where P is average power and A is root mean square (RMS) amplitude (for example,
typically, RMS voltage). Both signal and noise power (or amplitude) must be measured at
the same or equivalent points in a system, and within the same system bandwidth.

Because many signals have a very wide dynamic range, SNRs are usually expressed in
terms of the logarithmic decibel scale. In decibels, the SNR is, by definition, 10 times the
logarithm of the power ratio. If the signal and the noise is measured across the same
impedance then the SNR can be obtained by calculating 20 times the base-10 logarithm
of the amplitude ratio:

Electrical SNR and acoustics

Often the signals being compared are electromagnetic in nature, though it is also possible
to apply the term to sound stimuli. Due to the definition of decibel, the SNR gives the
same result independent of the type of signal which is evaluated (such as power, current,
or voltage).

Signal-to-noise ratio is closely related to the concept of dynamic range, where dynamic
range measures the ratio between noise and the greatest un-distorted signal on a channel.
SNR measures the ratio between noise and an arbitrary signal on the channel, not
necessarily the most powerful signal possible. Because of this, measuring signal-to-noise
ratios requires the selection of a representative or reference signal. In audio engineering,
this reference signal is usually a sine wave, sounding a tone, at a recognized and
standardized nominal level or alignment level, such as 1 kHz at +4 dBu (1.228 VRMS).

SNR is usually taken to indicate an average signal-to-noise ratio, as it is possible that

(near) instantaneous signal-to-noise ratios will be considerably different. The concept can
be understood as normalizing the noise level to 1 (0 dB) and measuring how far the signal
'stands out'. In general, higher signal to noise is better; the signal is 'cleaner'.

Image processing and interferometry

Main article: Signal to noise ratio (image processing)

In image processing, the SNR of an image is usually defined as the ratio of the mean
pixel value to the standard deviation of the pixel values, μ / σ (the inverse of the
coefficient of variation). Related measures are the "contrast ratio" and the "contrast-to-
noise ratio".

The connection between optical power and voltage in an imaging system is linear. This
usually means that the SNR of the electrical signal is calculated by the 10 log rule. With
an interferometric system, however, where interest lies in the signal from one arm only,
the field of the electromagnetic wave is proportional to the voltage (assuming that the
intensity in the second, the reference arm is constant). Therefore the optical power of the
measurement arm is directly proportional to the electrical power and electrical signals
from optical interferometry are following the 20 log rule.[1]

The Rose criterion (named after Albert Rose) states that an SNR of at least 5 is needed to
be able to distinguish image features at 100% certainty. An SNR less than 5 means less
than 100% certainty in identifying image details.[2]
For measurement devices in general

Recording of the noise of a thermogravimetric analysis device that is poorly isolated from
a mechanical point of view; the middle of the curve shows a lower noise, due to a lesser
surrounding human activity at night.

Any measurement device is disturbed by parasitic phenomena. This includes the

electronic noise as described above, but also any external event that affects the measured
phenomenon — wind, vibrations, gravitational attraction of the moon, variations of
temperature, variations of humidity etc. depending on what is measured and of the
sensitivity of the device.

It is often possible to reduce the noise by controlling the environment. Otherwise, when
the characteristics of the noise are known and are different from the signal's, it is possible
to filter it or to process the signal.

When the noise is a random perturbation and the signal is a constant value, it is possible
to enhance the SNR by increasing the measurement time.

Digital signals
When using digital storage the number of bits of each value determines the maximum
signal-to-noise ratio. In this case the noise is the error signal caused by the quantization
of the signal, taking place in the analog-to-digital conversion. The noise level is non-
linear and signal-dependent; different calculations exist for different signal models. The
noise is modeled as an analog error signal being summed with the signal before
quantization ("additive noise").

The modulation error ratio (MER) is a measure of the SNR in a digitally modulated
signal. Like SNR, MER can be expressed in dB.

Fixed point
See also: Fixed point arithmetic

For n-bit integers with equal distance between quantization levels (uniform quantization)
the dynamic range (DR) is also determined.
Assuming a uniform distribution of input signal values, the quantization noise is a
uniformly-distributed random signal with a peak-to-peak amplitude of one quantization
level, making the amplitude ratio 2n/1. The formula is then:

This relationship is the origin of statements like "16-bit audio has a dynamic range of 96
dB". Each extra quantization bit increases the dynamic range by roughly 6 dB.

Assuming a full-scale sine wave signal (that is, the quantizer is designed such that it has
the same minimum and maximum values as the input signal), the quantization noise
approximates a sawtooth wave with peak-to-peak amplitude of one quantization level[3]
and uniform distribution. In this case, the SNR is approximately

Floating point

Floating-point numbers provide a way to trade off signal-to-noise ratio for an increase in
dynamic range. For n bit floating-point numbers, with n-m bits in the mantissa and m bits
in the exponent:

Note that the dynamic range is much larger than fixed-point, but at a cost of a worse
signal-to-noise ratio. This makes floating-point preferable in situations where the
dynamic range is large or unpredictable. Fixed-point's simpler implementations can be
used with no signal quality disadvantage in systems where dynamic range is less than
6.02m. The very large dynamic range of floating-point can be a disadvantage, since it
requires more forethought in designing algorithms.[4]

Notes

• Analog-to-digital converters have other sources of noise that decrease the SNR
compared to the theoretical maximum from the idealized quantization noise.
• Often special filters are used to weight the noise: DIN-A, DIN-B, DIN-C, DIN-D,
CCIR-601; for video, special filters such as comb filters may be used.
• Maximum possible full scale signal can be charged as peak-to-peak or as RMS.
Audio uses RMS, Video P-P, which gave +9 dB more SNR for video.
• It is more common to express SNR in digital systems using Eb/No - the energy per
bit per noise power spectral density.

Further information: Quantization noise, Bit resolution

Informal use
Informally, "signal-to-noise ratio" refers to the ratio of useful information to false or
irrelevant data

Eb/N0 (the energy per bit to noise power spectral density ratio) is an important
parameter in digital communication or data transmission. It is a normalized signal-to-
noise ratio (SNR) measure, also known as the "SNR per bit". It is especially useful when
comparing the bit error rate (BER) performance of different digital modulation schemes
without taking bandwidth into account.

Eb/N0 is equal to the SNR divided by the "gross" link spectral efficiency in (bit/s)/Hz,
where the bits in this context are transmitted data bits, inclusive of error correction
information and other protocol overhead. It should be noted that when forward error
correction is being discussed, Eb/N0 is routinely used to refer to the energy per
information bit (i.e. the energy per bit net of FEC overhead bits)[citation needed]. In this
context, Es/N0 is generally used to relate actual transmitted power to noise[clarification needed].

The noise spectral density N0, usually expressed in units of watts per hertz, can also be
seen as having dimensions of energy, or units of joules, or joules per cycle. Eb/N0 is
therefore a non-dimensional ratio.

Eb/N0 is commonly used with modulation and coding designed for noise-limited rather
than interference-limited communication, and for power-limited rather than bandwidth-
limited communications.[clarification needed] Examples of power-limited communications
include deep-space and spread spectrum, and is optimized by using large bandwidths
relative to the bit rate.

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Relation to carrier-to-noise ratio

Eb/N0 is closely related to the carrier-to-noise ratio (CNR or C/N), i.e. the signal-to-noise
ratio (SNR) of the received signal, after the receiver filter but before detection:

where
 fb is the channel data rate (net bitrate), and
B is the channel bandwidth

The equivalent expression in logarithmic form (dB):

Caution: Sometimes, the noise power is denoted by N0 / 2 when negative frequencies and
complex-valued equivalent baseband signals are considered, and in that case, there will
be a 3dB difference.

Relation to Es/N0
Eb/N0 can be seen as a normalized measure of the energy per symbol per noise power
spectral density (Es/N0), where Es is the Energy per symbol in Joules. This measure is
also commonly used in the analysis of digital modulation schemes. The two quotients are
related to each other according to the following:

where

M is the number of alternative modulation symbols.

Es/N0 can further be expressed as:

where

C/N is the carrier-to-noise ratio or signal-to-noise ratio.

B is the channel bandwidth in Hertz.
fs is the symbol rate in baud or symbols/second.

For a PSK, ASK or QAM modulation with pulse shaping such as raised cosine shaping,
the B/fs ratio is usually slightly larger than 1, depending of the pulse shaping filter.

Shannon limit
The Shannon–Hartley theorem says that the limit of reliable data rate of a channel
depends on bandwidth and signal-to-noise ratio according to:
where

R is an information rate in bits per second;

B is the bandwidth of the channel in hertz;
S is the total signal power (equivalent to the carrier power C); and
N is the total noise power in the bandwidth.

This equation can be used to establish a bound on Eb/N0 for any system that achieves
reliable communication, by considering a bit rate equal to R and therefore an average
energy per bit of Eb = S/R, with noise spectral density of N0 = N/B. For this calculation, it
is conventional to define a normalized rate Rl = R/(2B), a bandwidth utilization parameter
of bits per second per half hertz, or bits per dimension (a signal of bandwidth B can be
encoded with 2B dimensions, according to the Nyquist–Shannon sampling theorem).
Making appropriate substitutions, the Shannon limit is:

Which can be solved to get the Shannon-limit bound on Eb/N0:

When the data rate is small compared to the bandwidth, so that Rl is near zero, the bound,
sometimes called the ultimate Shannon limit,[1] is:

which corresponds to –1.59 dB.

Cutoff rate
For any given system of coding and decoding, there exists what is known as a cutoff rate
R0, typically corresponding to an Eb/N0 about 2 dB above the Shannon capacity limit.
[citation needed]
The cutoff rate used to be thought of as the limit on practical error correction
codes without an unbounded increase in processing complexity, but has been rendered
largely obsolete by the more recent discovery of turbo codes.

References
 1. ^ Nevio Benvenuto and Giovanni Cherubini (2002). Algorithms for
Communications Systems and Their Applications. John Wiley & Sons. p. 508.
ISBN 0470843896.

External links
• Eb/N0 Explained. An introductory article on Eb/N0

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Symbol rate
From Wikipedia, the free encyclopedia
Jump to: navigation, search
It has been suggested that this article or section be merged with Baud. (Discuss)

In digital communications, symbol rate, also known as baud or modulation rate; is the
number of symbol changes (signalling events) made to the transmission medium per
second using a digitally modulated signal or a line code. The Symbol rate is measured in
baud (Bd) or symbols/second. In the case of a line code, the symbol rate is the pulse rate
in pulses/second. Each symbol can represent or convey one or several bit of data. The
symbol rate is related to but should not be confused with gross bitrate expressed in bit/s.

Contents
[show]

[edit] Symbols
A symbol is a state or significant condition of the communication channel that persists for
a fixed period of time. A sending device places symbols on the channel at a fixed and
known symbol rate,and the receiving device has the job of detecting the sequence of
symbols in order to reconstruct the transmitted data. There may be a direct
correspondence between a symbol and a small unit of data (for example, each symbol
may encode one or several binary bits) or the data may be represented by the transitions
between symbols or even by a sequence of many symbols.
The symbol duration time, also known as unit interval, can be directly measured as the
time between transitions by looking into an eye diagram of an oscilloscope. The symbol
duration time Ts can be calculated as:

where fs is the symbol rate.

A simple example: A baud rate of 1 kBd = 1,000 Bd is synonymous to a symbol

rate of 1,000 symbols per second. In case of a modem, this corresponds to 1,000
tones per second, and in case of a line code, this corresponds to 1,000 pulses per
second. The symbol duration time is 1/1,000 second = 1 millisecond.

[edit] Relationship to gross bitrate

The term baud rate has sometimes incorrectly been used to mean bit rate, since these rates
are the same in old modems as well as in the simplest digital communication links using
only one bit per symbol, such that binary "0" is represented by one symbol, and binary
"1" by another symbol. In more advanced modems and data transmission techniques, a
symbol may have more than two states, so it may represent more than one binary bit (a
binary bit always represents exactly two states). For this reason, the baud rate value will
often be lower than the gross bit rate.

Example of use and misuse of "baud rate": It is correct to write "the baud rate of my
COM port is 9,600" if we mean that the bit rate is 9,600 bit/s, since there is one bit per
symbol in this case. It is not correct to write "the baud rate of Ethernet is 100 Mbaud" or
"the baud rate of my modem is 56,000" if we mean bit rate. See below for more details on
these techniques. The difference between baud (or signalling rate) and the data rate (or bit
rate) is like a man using a single semaphore flag who can move his arm to a new position
once each second, so his signalling rate (baud) is one symbol per second. The flag can be
held in one of eight distinct positions: Straight up, 45° left, 90° left, 135° left, straight
down (which is the rest state, where he is sending no signal), 135° right, 90° right, and
45° right. Each signal carries three bits of information. It takes three binary digits to
encode eight states. The data rate is three bits per second. In the Navy, more than one flag
pattern and arm can be used at once, so the combinations of these produce many symbols,
each conveying several bits, a higher data rate.

If N bits are conveyed per symbol, and the gross bit rate is R, inclusive of channel coding
overhead, the symbol rate can be calculated as:

In that case M=2N different symbols are used. In a modem, these may be sinewave tones
with unique combinations of amplitude, phase and/or frequency. For example, in a
64QAM modem, M=64. In a line code, these may be M different voltage levels.

By taking information per pulse N in bit/pulse to be the base-2-logarithm of the number

of distinct messages M that could be sent, Hartley[1] constructed a measure of the gross
bitrate R as:
where fs is the baud rate in symbols/second or pulses/second. (See Hartley's law).

[edit] Modems for passband transmission

Modulation is used in passband filtered channels such as telephone lines, radio channels
and other frequency division multiplex (FDM) channels.

In a digital modulation method provided by a modem, each symbol is typically a sine

wave tone with certain frequency, amplitude and phase. The baud rate is the number of
transmitted tones per second.

One symbol can carry one or several bits of information. In voiceband modems for the
telephone network, it is common for one symbol to carry up to 10 bits per symbol.

Conveying more than one bit per symbol or bit per pulse has advantages. It reduces the
time required to send a given quantity of data over a limited bandwidth. A high spectral
efficiency in (bit/s)/Hz can be achieved, i.e. a high bit rate in bit/s although the bandwidth
in hertz may be low.

The maximum baud rate for a passband for common modulation methods such as QAM,
PSK and OFDM is approximately equal to the passband bandwidth.

Voiceband modem examples:

• A V.22bis modem transmits 2400 bit/s using 1200 Bd (1200 symbol/s), where
each quadrature amplitude modulation symbol carries two bits of information.
The modem can generate M=22=4 different symbols. It requires a bandwidth of
2400 Hz (equal to the baud rate). The carrier frequency (the central frequency of
the generated spectrum) is 1800 Hz, meaning that the lower cut off frequency is
1800 -2400/2 = 600 Hz, and the upper cutoff frequency is 1800 + 2400/2 = 3000
Hz.

• A V.34 modem may transmit symbols at a baud rate of 3,420 Bd, and each
symbol can carry up to ten bits, resulting in a gross bit rate of 3420 * 10 = 34,200
bit/s. However, the modem is said to operate at a net bit rate of 33,800 bit/s,
excluding physical layer overhead.

[edit] Line codes for baseband transmission

In case of a baseband channel such as a telegraph line, a serial cable or a Local Area
Network twisted pair cable, data is transferred using line codes, i.e. pulses rather than
sinewave tones. In this case the baud rate is synonymous to the pulse rate in
pulses/second.

The maximum baud rate or pulse rate for a base band channel is called the Nyquist rate,
and is half the bandwidth (half the upper cut-off frequency).
The simplest digital communication links (such as individual wires on a motherboard or
the RS-232 serial port/COM port) typically have a symbol rate equal to the gross bit rate.

Common communication links such as 10 Mbit/s Ethernet (10Base-T), USB, and

FireWire typically have a symbol rate slightly higher than the data bit rate, due to the
overhead of extra non-data symbols used for self-synchronizing code and error detection.

J. M. Emile Baudot (1845–1903) worked out a five-level code (five bits per character) for
telegraphs which was standardized internationally and is commonly called Baudot code.

More than two voltage levels are used in advanced techniques such as FDDI and
100/1000 Mbit/s Ethernet LANs, and others, to achieve high data rates.

1000 Mbit/s Ethernet LAN cables use many wire pairs and many bits per symbol to
encode their data payloads. 1000BASE-T uses four wire pairs and two data bits per
symbol to get a symbol rate of 125 MBd (i.e., 1.25×108 Bd).

[edit] Digital television and OFDM example

In digital television transmission the symbol rate calculation is:

symbol rate in symbols per second = (Data rate in bits per second * 204) / (188 *
bits per symbol)

The 204 is the number of bytes in a packet including the 16 trailing Reed-Solomon error
checking and correction bytes. The 188 is the number of data bytes (187 bytes) plus the
leading packet sync byte (0x47).

The bits per symbol is the (modulation's power of 2)*(Forward Error Correction). So for
example in 64-QAM modulation 64 = 26 so the bits per symbol is 6. The Forward Error
Correction (FEC) is usually expressed as a fraction, i.e., 1/2, 3/4, etc. In the case of 3/4
FEC, for every 3 bits of data, you are sending out 4 bits, one of which is for error
correction.

Example:

given bit rate = 18096263

Modulation type = 64-QAM
FEC = 3/4

then

In digital terrestrial digital television (DVB-T, DVB-H and similar techniques) OFDM
modulation is used, i.e. multi-carrier modulation. The above symbol rate should then be
divided by the number of OFDM sub-carriers in view to achieve the OFDM symbol rate.
See the OFDM system comparison table for further numerical details.
[edit] Relationship to chip rate

Some communication links (such as GPS transmissions, CDMA cell phones, and other
spread spectrum links) have a symbol rate much higher than the data rate (they transmit
many symbols called chips per data bit. Representing one bit by a chip sequence of many
symbols overcomes co-channel interference from other transmitters sharing the same
frequency channel, including radio jamming, and is common in military radio and cell
phones. Despite the fact that using more bandwidth to carry the same bit rate gives low
channel spectral efficiency in (bit/s)/Hz, it allows many simultaneous users, which results
in high system spectral efficiency in (bit/s)/Hz per unit of area.

In these systems, the symbol rate of the physically transmitted high-frequency signal rate
is called chip rate, which also is the pulse rate of the equivalent base band signal.
However, in spread spectrum systems, the term symbol may also be used at a higher layer
and refer to one information bit, or a block of information bits that are modulated using
for example conventional QAM modulation, before the CDMA spreading code is
applied. Using the latter definition, the symbol rate is equal to or lower than the bit rate.

[edit] Relationship to bit error rate

The disadvantage of conveying many bits per symbol is that the receiver has to
distinguish many signal levels or symbols from each other, which may be difficult and
cause bit errors in case of a poor phone line that suffers from low signal-to-noise ratio. In
that case, a modem or network adapter may automatically choose a slower and more
robust modulation scheme or line code, using fewer bits per symbol, in view to reduce
the bit error rate.

An optimal symbol set design takes into account channel bandwidth, desired information
rate, noise characteristics of the channel and the receiver, and receiver and decoder
complexity.

[edit] Modulation
Many data transmission systems operate by the modulation of a carrier signal. For
example, in frequency-shift keying (FSK), the frequency of a tone is varied among a
small, fixed set of possible values. In a synchronous data transmission system, the tone
can only be changed from one frequency to another at regular and well-defined intervals.
The presence of one particular frequency during one of these intervals constitutes a
symbol. (The concept of symbols does not apply to asynchronous data transmission
systems.) In a modulated system, the term modulation rate may be used synonymously
with symbol rate.
[edit] Binary Modulation

If the carrier signal has only two states, then only one bit of data (i.e., a 0 or 1) can be
transmitted in each symbol. The bit rate is in this case equal to the symbol rate. For
example, a binary FSK system would allow the carrier to have one of two frequencies,
one representing a 0 and the other a 1. A more practical scheme is differential binary
phase-shift keying, in which the carrier remains at the same frequency, but can be in one
of two phases. During each symbol, the phase either remains the same, encoding a 0, or
jumps by 180°, encoding a 1. Again, only one bit of data (i.e., a 0 or 1) is transmitted by
each symbol. This is an example of data being encoded in the transitions between
symbols (the change in phase), rather than the symbols themselves (the actual phase).
(The reason for this in phase-shift keying is that it is impractical to know the reference
phase of the transmitter.)

[edit] N-ary Modulation, N greater than 2

By increasing the number of states that the carrier signal can take, the number of bits
encoded in each symbol can be greater than one. The bit rate can then be greater than the
symbol rate. For example, a differential phase-shift keying system might allow four
possible jumps in phase between symbols. Then two bits could be encoded at each
symbol interval, achieving a data rate of double the symbol rate. In a more complex
scheme such as 16-QAM, four bits of data are transmitted in each symbol, resulting in a
bit rate of four times the symbol rate.

[edit] Data Rate versus Error Rate

Modulating a carrier increases the frequency range, or bandwidth, it occupies.

Transmission channels are generally limited in the bandwidth they can carry. The
bandwidth depends on the symbol (modulation) rate (not directly on the bit rate). As the
bit rate is the product of the symbol rate and the number of bits encoded in each symbol,
it is clearly advantageous to increase the latter if the former is fixed. However, for each
additional bit encoded in a symbol, the constellation of symbols (the number of states of
the carrier) doubles in size. This makes the states less distinct from one another which in
turn makes it more difficult for the receiver to detect the symbol correctly in the presence
of disturbances on the channel.

The history of modems is the attempt at increasing the bit rate over a fixed bandwidth
(and therefore a fixed maximum symbol rate), leading to increasing bits per symbol. For
example, the V.29 specifies 4 bits per symbol, at a symbol rate of 2,400 baud, giving an
effective bit rate of 9,600 bits per second.

The history of spread spectrum goes in the opposite direction, leading to fewer and fewer
data bits per symbol in order to spread the bandwidth. In the case of GPS, we have a data
rate of 50 bit/s and a symbol rate of 1.023 Mchips/s. If each chip is considered a symbol,
each symbol contains far less than one bit ( 50 bit/s / 1023 Ksymbols/s =~= 0.000 05
bits/symbol ).
The complete collection of M possible symbols over a particular channel is called a M-
ary modulation scheme. Most modulation schemes transmit some integer number of bits
per symbol b, requiring the complete collection to contain M = 2^b different symbols.
Most popular modulation schemes can be described by showing each point on a
constellation diagram, although a few modulation schemes (such as MFSK, DTMF,
pulse-position modulation, spread spectrum modulation) require a different description.

[edit] Significant condition

In telecommunication, in the modulation of a carrier, a significant condition is one of the
values of the signal parameter chosen to represent information.[2]

A significant condition could be an electrical current (voltage, or power level), an optical

power level, a phase value, or a particular frequency or wavelength. The duration of a
significant condition is the time interval between successive significant instants.[2] A
change from one significant condition to another is called a signal transition. Information
can be transmitted either during the given time interval, or encoded as the presence or
absence of a change in the received signal.[3]

Significant conditions are recognized by an appropriate device called a receiver,

demodulator, or decoder. The decoder translates the actual signal received into its
intended logical value such as a binary digit (0 or 1), an alphabetic character, a mark, or a
space. Each significant instant is determined when the appropriate device assumes a
condition or state usable for performing a specific function, such as recording,
processing, or gating.[2]

[edit] References
1. ^ D. A. Bell (1962). Information Theory; and its Engineering Applications (3rd ed.). New York: Pitman.
2. ^ a b c Federal Standard 1037C. National Communications System. 1996-07-07. http://www.its.bldrdoc.gov/fs-1037/fs-
1037c.htm.
3. ^ "System Design and Engineering Standard for Tactical Communications". Mil-Std-188-200 (United States
Department of Defense). 1983-05-28. http://assist.daps.dla.mil/quicksearch/basic_profile.cfm?ident_number=35582.

[edit] See also

• Chip rate
• Gross bit rate, also known as data signaling rate or line rate.
• bandwidth
• Bitrate
• Constellation diagram, which shows (on a graph or 2D oscilloscope image) how a
given signal state (a symbol) can represent three or four bits at once.
• List of device bandwidths
 • PCM

[edit] External links

• What is the Symbol rate?
• "On the origins of serial communications and data encoding".
http://www.compkarori.com/dbase/bu07sh.htm. Retrieved on January 4 2007.

Retrieved from "http://en.wikipedia.org/wiki/Symbol_rate"

Categories: Data transmission
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A set of techniques where by a sequence of information-carrying quantities occurring

at discrete instances of time is encoded into a corresponding regular sequence of
electromagnetic carrier pulses. Varying the amplitude, polarity, presence or absence,
duration, or occurrence in time of the pulses gives rise to the four basic forms of pulse
modulation: pulse-amplitude modulation (PAM), pulse-code modulation (PCM), pulse-
width modulation (PWM, also known as pulse-duration modulation, PDM), and pulse-
position modulation (PPM).

Analog-to-digital conversion

An important concept in pulse modulation is analog-to-digital (A/D) conversion, in which

an original analog (time- and amplitude-continuous) information signal s(t) is changed at
the transmitter into a series of regularly occurring discrete pulses whose amplitudes are
restricted to a fixed and finite number of values. An inverse digital-to-analog (D/A)
process is used at the receiver to reconstruct an approximation of the original form of s(t).
Conceptually, analog-to-digital conversion involves two steps. First, the range of
amplitudes of s(t) is divided or quantized into a finite number of predetermined levels,
and each such level is represented by a pulse of fixed amplitude. Second, the amplitude
of s(t) is periodically measured or sampled and replaced by the pulse representing the
level that corresponds to the measurement. See also Analog-to-digital converter; Digital-
to-analog converter.

According to the Nyquist sampling theorem, if sampling occurs at a rate at least twice
that of the bandwidth of s(t), the latter can be unambiguously reconstructed from its
amplitude values at the sampling instants by applying them to an ideal low-pass filter
whose bandwidth matches that of s(t).

Quantization, however, introduces an irreversible error, the so-called quantization error,

since the pulse representing a sample measurement determines only the quantization level
in which the measurement falls and not its exact value. Consequently, the process of
reconstructing s(t) from the sequence of pulses yields only an approximate version of s(t).

Pulse-amplitude modulation

In PAM the successive sample values of the analog signal s(t) are used to effect the
amplitudes of a corresponding sequence of pulses of constant duration occurring at the
sampling rate. No quantization of the samples normally occurs (Fig. 1a, b). In principle
the pulses may occupy the entire time between samples, but in most practical systems the
pulse duration, known as the duty cycle, is limited to a fraction of the sampling interval.
Such a restriction creates the possibility of interleaving during one sample interval one or
more pulses derived from other PAM systems in a process known as time-division
multiplexing (TDM). See also Multiplexing and multiple access.
 sine wave. (a) Analog signal, s(t). (b) Pulse-
amplitude modulation. (c) Pulse-width modulation. (d) Pulse-position modulation.">
Forms of pulse modulation for the case where the analog signal, s(t), is a sine wave. (a)
Analog signal, s(t). (b) Pulse-amplitude modulation. (c) Pulse-width modulation. (d)
Pulse-position modulation.

Pulse-width modulation

In PWM the pulses representing successive sample values of s(t) have constant
amplitudes but vary in time duration in direct proportion to the sample value. The pulse
duration can be changed relative to fixed leading or trailing time edges or a fixed pulse
center. To allow for time-division multiplexing, the maximum pulse duration may be
limited to a fraction of the time between samples (Fig. 1c).

Pulse-position modulation

PPM encodes the sample values of s(t) by varying the position of a pulse of constant
duration relative to its nominal time of occurrence. As in PAM and PWM, the duration of
the pulses is typically a fraction of the sampling interval. In addition, the maximum time
excursion of the pulses may be limited (Fig. 1d).

Pulse-code modulation

Many modern communication systems are designed to transmit and receive only pulses
of two distinct amplitudes. In these so-called binary digital systems, the analog-to-digital
conversion process is extended by the additional step of coding, in which the amplitude
of each pulse representing a quantized sample of s(t) is converted into a unique sequence
of one or more pulses with just two possible amplitudes. The complete conversion
process is known as pulse-code modulation.

Figure 2a shows the example of three successive quantized samples of an analog signal
s(t), in which sampling occurs every T seconds and the pulse representing the sample is
limited to T/2 seconds. Assuming that the number of quantization levels is limited to 8,
each level can be represented by a unique sequence of three two-valued pulses. In Fig. 2b
these pulses are of amplitude V or 0, whereas in Fig. 2c the amplitudes are V and −V.

Pulse-code modulation. (a) Three successive quantized samples of an analog signal. (b)
With pulses of amplitude V or 0. (c) With pulses of amplitude V or −V.

PCM enjoys many important advantages over other forms of pulse modulation due to the
fact that information is represented by a two-state variable. First, the design parameters of
a PCM transmission system depend critically on the bandwidth of the original signal s(t)
and the degree of fidelity required at the point of reconstruction, but are otherwise largely
independent of the information content of s(t). This fact creates the possibility of
deploying generic transmission systems suitable for many types of information. Second,
the detection of the state of a two-state variable in a noisy environment is inherently
simpler than the precise measurement of the amplitude, duration, or position of a pulse in
which these quantities are not constrained. Third, the binary pulses propagating along a
medium can be intercepted and decoded at a point where the accumulated distortion and
attenuation are sufficiently low to assure high detection accuracy. New pulses can then be
generated and transmitted to the next such decoding point. This so-called process of
repeatering significantly reduces the propagation of distortion and leads to a quality of
transmission that is largely independent of distance.

Time-division multiplexing
An advantage inherent in all pulse modulation systems is their ability to transmit signals
from multiple sources over a common transmission system through the process of time-
division multiplexing. By restricting the time duration of a pulse representing a sample
value from a particular analog signal to a fraction of the time between successive
samples, pulses derived from other sampled analog signals can be accommodated on the
transmission system.

One important application of this principle occurs in the transmission of PCM telephone
voice signals over a digital transmission system known as a T1 carrier. In standard T1
coding, an original analog voice signal is band-limited to 4000 hertz by passing it through
a low-pass filter, and is then sampled at the Nyquist rate of 8000 samples per second, so
that the time between successive samples is 125 microseconds. The samples are
quantized to 256 levels, with each of them being represented by a sequence of 8 binary
pulses. By limiting the duration of a single pulse to 0.65 microsecond, a total of 193
pulses can be accommodated in the time span of 125 microseconds between samples.
One of these serves as a synchronization marker that indicates the beginning of such a
sequence of 193 pulses, while the other 192 pulses are the composite of 8 pulses from
each of 24 voice signals, with each 8-pulse sequence occupying a specified position. T1
carriers and similar types of digital carrier systems are in widespread use in the world's
telephone networks.

Bandwidth requirements

Pulse modulation systems may incur a significant bandwidth penalty compared to the
transmission of a signal in its analog form. An example is the standard PCM transmission
of an analog voice signal band-limited to 4000 hertz over a T1 carrier. Since the
sampling, quantizing, and coding process produces 8 binary pulses 8000 times per second
for a total of 64,000 binary pulses per second, the pulses occur every 15.625
microseconds. Depending on the shape of the pulses and the amount of intersymbol
interference, the required transmission bandwidth will fall in the range of 32,000 to
64,000 hertz. This compares to a bandwidth of only 4000 hertz for the transmission of the
signal in analog mode. See also Bandwidth requirements (communications).

Applications

PAM, PWM, and PPM found significant application early in the development of digital
communications, largely in the domain of radio telemetry for remote monitoring and
sensing. They have since fallen into disuse in favor of PCM.

Since the early 1960s, many of the world's telephone network providers have gradually,
and by now almost completely, converted their transmission facilities to PCM
technology. The bulk of these transmission systems use some form of time-division
multiplexing, as exemplified by the 24-voice channel T1 carrier structure. These carrier
systems are implemented over many types of transmission media, including twisted pairs
of telephone wiring, coaxial cables, fiber-optic cables, and microwave. See also Coaxial
cable; Communications cable; Microwave; Optical communications; Optical fibers;
Switching systems (communications).

The deployment of high-speed networks such as the Integrated Service Digital Network
(ISDN) in many parts of the world has also relied heavily on PCM technology. PCM and
various modified forms such as delta modulation (DM) and adaptive differential pulse-
code modulation (ADPCM) have also found significant application in satellite
transmission systems