Digital Signal

Pprocessing : DDS
Dilip K. Saha
Direct Digital Synthesis (DDS)
 What is Direct Digital Synthesis?
Direct digital synthesis (DDS) is a method of
producing an analog waveform—usually a sine
wave—by generating a time-varying signal in
digital form and then performing a digital-to-
analog conversion.
 Because operations within a DDS device are
primarily digital, it can offer fast switching
between output frequencies, fine frequency
resolution, and operation over a broad
spectrum of frequencies.
 With advances in design and process
technology, today’s DDS devices are very
compact and draw little power.
 Why would one use a direct digital synthesizer
(DDS)? Aren’t there other methods for easily
generating frequencies?
• The ability to accurately produce and control waveforms of
various frequencies and profiles has become a key requirement
common to a number of industries.
• DDS technique is rapidly gaining acceptance for solving
frequency- (or waveform) generation requirements in both
communications and industrial applications because single-chip
IC devices can generate programmable analog output
waveforms simply and with high resolution and accuracy.
• Furthermore, the cost and power consumption levels are very
low. For example, the AD9833, a DDS-based programmable
waveform generator (Figure 1), operating at 5.5 V with a 25-
MHz clock, consumes a maximum power of 30 milliwatts.

Figure 1. The AD9833—a one-

chip waveform generator.
 What are the main benefits of using a
DDS?
• DDS devices like the AD9833 are programmed through a
high speed serial peripheral-interface (SPI), and need
only an external clock to generate simple sine waves.
DDS devices are now available that can generate
frequencies from less than 1 Hz up to 400 MHz (based
on a 1-GHz clock).
• The benefits of their low power, low cost, and single
small package, combined with their inherent excellent
performance and the ability to digitally program (and
reprogram) the output waveform, make DDS devices an
extremely attractive solution—preferable to less-flexible
solutions comprising aggregations of discrete elements.
 What kind of outputs can I generate with a
typical DDS device?
• DDS devices are not limited
to purely sinusoidal outputs.
Figure 2 shows the square-,
triangular-, and sinusoidal
outputs available from an
AD9833.

Figure 2. Square-, triangular-,

and sinusoidal outputs from
a DDS.
How does a DDS device create a sine
wave?
Here’s a breakdown of the internal circuitry of a DDS device: its main
components are a phase accumulator, a means of phase-to-amplitude
conversion (often a sine look-up table), and a DAC. These blocks are
represented in Figure 3.

Figure 3.
Components of a
direct digital
synthesizer.

A DDS produces a sine wave at a given frequency. The frequency

depends on two variables, the reference-clock frequency and the binary
number programmed into the frequency register (tuning word).
• The binary number in the frequency register provides
the main input to the phase accumulator. If a sine look-
up table is used, the phase accumulator computes a
phase (angle) address for the look-up table, which
outputs the digital value of amplitude—corresponding
to the sine of that phase angle—to the DAC. The DAC,
in turn, converts that number to a corresponding value
of analog voltage or current.
• To generate a fixed-frequency sine wave, a constant
value (the phase increment—which is determined by
the binary number) is added to the phase accumulator
with each clock cycle.
• If the phase increment is large, the phase accumulator
will step quickly through the sine look-up table and thus
generate a high frequency sine wave. If the phase
increment is small, the phase accumulator will take
many more steps, accordingly generating a slower
waveform.
What do you mean by a complete
DDS?
•
The integration of a D/A converter and a
DDS onto a single chip is commonly
known as a complete DDS solution, a
property common to all DDS devices from
ADI.
Analogue to Digital
Conversion (ADC)
Dilip K. Saha
 ADC Analogue to Digital Conversion
 In a digital system the original analogue audio signal is converted
into binary coded digital signal, called word. The process consists of
:
 Sampling– taking samples of the analogue waveform at a fixed rate
which must be at least twice the highest audio frequency used. Each
sample is placed in the memory and at the same time measured
and converted into a code. For example, the sampling rate for
audio, recommended by AES are:
48 KHz---- for professional applications
44 KHz---- for commercial applications
 Quantizing– It is process of measuring and comparing the samples
with a standard scale of quantized level of discrete numbers.
 Double coding--- The quantize levels are converted into a binary
code representing a group of number one byte (or 8 bits), 2 bytes
(16 bits), etc. Any error generated is corrected by ring parity
generator circuits. A binary code of 10 bits generates 60 dB S/N
ratio, 13 bits 70 dB, etc. For higher S/N ratio higher number of bits 4
bytes to 30 bytes are usually selected. Following Figures show the
ADC & DAC processes.
Digital Modulation

Dilip K. Saha
 Modulation
In telecommunications, modulation is the process of varying
some characteristic of an r.f. Waveform, called carrier in
accordance with the instantaneous value of a low frequency
information signal, called modulated signal.

Analog modulation methods

In analog modulation, the modulation is applied continuously in
response to the analog information signal.Common analog
modulation techniques are:
Amplitude modulation (AM)
Frequency modulation (FM)
Phase modulation (PM)
 Digital modulation
• Digital modulation (also referred to as shift
keying) is a modulation in which an analog
carrier signal is modulated by a digital bit
stream.
• In digital modulation, the modified parameter
of the carrier signal can take only discrete
values.
• This type of modulation can sometimes be
referred to also as discrete modulation or
manipulation.
The most common digital modulation techniques are:

Amplitude-shift keying (ASK)

Frequency-shift keying (FSK)

Phase-shift keying (PSK)

Quadrature amplitude modulation (QAM) a combination

of PSK and ASK

Orthogonal frequency division multiplexing (OFDM)

 Amplitude-shift keying (ASK)
• It is a form of modulation that conveys digital data by
varying the amplitude of a carrier wave.
• The amplitude of an analog carrier signal varies in
accordance with the bit stream (modulating signal),
keeping frequency and phase constant.
• The level of amplitude can be used to represent binary
logic 0s and 1s. We can think of a carrier signal as an ON
or OFF switch. In the modulated signal, logic 0 is
represented by the absence of a carrier, thus giving
OFF/ON keying operation and hence the name given.
• Like AM, ASK is also linear and sensitive to atmospheric
noise, distortions, propagation conditions on different
routes in PSTN, etc. It requires excessive bandwidth and
is therefore a waste of power. Both ASK modulation and
demodulation processes are relatively inexpensive. The
ASK technique is also commonly used to transmit digital
data over optical fiber.
An example of
binary FSK

• Frequency-shift keying (FSK) is a form of

frequency modulation in which the modulating signal shifts the
output frequency between predetermined values.
• Usually, the instantaneous frequency is shifted between two
discrete values termed the mark frequency and the space
frequency
 Phase-shift keying (PSK)
• It is a digital modulation scheme that conveys data by
changing, or modulating, the phase of a reference signal
(the carrier wave).
• A convenient way to represent PSK schemes is on a
constellation diagram, Where the real and imaginary axes
are termed the in-phase and quadrature axes respectively
due to their 90° separation. The amplitude of each point
along the in-phase axis is used to modulate a cosine (or
sine) wave and the amplitude along the quadrature axis to
modulate a sine (or cosine) wave.
• In PSK, the constellation points chosen are usually
positioned with uniform angular spacing around a circle.
This gives maximum phase-separation between adjacent
points and thus the best immunity to corruption. Two
common examples are binary phase-shift keying (BPSK)
which uses two phases, and
quadrature phase-shift keying (QPSK) which uses four
phases, although any number of phases may be used.
Constellation diagram for Constellation diagram for
BPSK QPSK
Bitrate
In telecommunications and computing, bitrate is the number of bits
that are conveyed or processed per unit of time.
The bit rate is quantified using the 'bit per second' (bit/s or bps) unit,
often in conjunction with a SI prefix such as kilo (kbit/s or kbps), Mega
(Mbit/s or Mbps), Giga (Gbit/s or Gbps) or Tera (Tbit/s or Tbps).

Symbol (data)
In digital communications, a symbol is the smallest unit of data transmitted
at one time.
In the simplest modulation schemes such as binary phase-shift keying, only
one bit of data (i.e., a 0 or 1) is transmitted at a time depending on the
phase of the transmitted signal. However, in a more complex scheme such
as 16-QAM, four bits of data are transmitted simultaneously, resulting in a
symbol rate (or baud rate) that is equal to one quarter of the bit rate.
Symbol rate
In digital communications, the symbol rate is the bit rate divided by the
number of bits transmitted in each symbol. Symbol rate is measured in
symbols-per-second, hertz (Hz), or baud (Bd).
Orthogonal frequency-division
multiplexing
• Orthogonal Frequency-Division Multiplexing (OFDM) — essentially
the same thing as Coded OFDM (COFDM) as well as discrete multi-
tone modulation (DMT) — is based upon the principle of
frequency-division multiplexing (FDM), but is implemented as a digital
modulation scheme. The bit stream to be transmitted is split into
several parallel bit streams, typically dozens to thousands. The
available frequency spectrum is divided into several sub-channels,
and each low-rate bit stream is transmitted over one sub-channel by
modulating a sub-carrier using a standard modulation scheme, for
example PSK, QAM, etc. The sub-carrier frequencies are chosen so
that the modulated data streams are orthogonal to each other,
meaning that cross-talk between the sub-channels is eliminated. This
orthogonality occurs when sub-carriers are equally spaced by the
symbol rate of a sub-carrier.
• Channel equalization is simplified by using many slowly modulated
narrowband signals instead of one rapidly modulated wideband signal.
The primary advantage of OFDM is its ability to cope with severe
channel conditions — for example, multipath and narrowband
interference — without complex equalization filters.
Transmitter
An OFDM carrier signal is the sum of a number of
orthogonal sub-carriers, with baseband data on each sub-
carrier being independently modulated commonly using
some type of quadrature amplitude modulation (QAM) or
phase-shift keying (PSK). This composite baseband signal
is typically used to modulate a main RF carrier.
Receiver
The receiver picks up the signal r(t), which is then quadrature-mixed down
to base band using cosine and sine waves at the carrier frequency. This
also creates signals centered on 2fc, so low-pass filters are used to reject
these. The base band signals are then sampled and digitized using
analogue-to-digital converters (ADCs), and a forward FFT is used to
convert back to the frequency domain.
This returns N parallel streams, each of which is converted to a binary
stream using an appropriate symbol detector. These streams are then re-
combined into a serial stream, , which is an estimate of the original binary
stream at the transmitter.
 Usages
OFDM has developed into a popular scheme for wideband
digital communication systems. Examples of applications are:
• ADSL and VDSL broadband access via telephone network
copper wires.
• IEEE 802.11a and 802.11g Wireless LANs.
• The Digital audio broadcasting systems EUREKA 147,
Digital Radio Mondiale, HD Radio, T-DMB and ISDB-TSB.
• Terrestrial digital TV systems DVB-T, DVB-H, T-DMB and
ISDB-T.
• IEEE 802.16 or WiMax Wireless MAN.
• IEEE 802.20 or Mobile Broadband Wireless Access (MBWA).
• Flash-OFDM cellular system.
• Some Ultra wideband (UWB) systems.
• Power line communication (PLC).
• Other Point-to-point (PtP) and point-to-multipoint (PtMP)
wireless applications
 Usage
• Terrestrial digital radio and television broadcasting.
Much of Europe and Asia has adopted OFDM for terrestrial
broadcasting of digital television (DVB-T, DVB-H and T-DMB) and
radio (EUREKA 147 DAB, Digital Radio Mondiale, HD Radio and
T-DMB).
• Powerline Technology
OFDM is used by powerline devices to extend Ethernet connections
to other rooms in a home through its power wiring.
Adaptive modulation is particularly important with such a noisy
channel as electrical wiring.
• ADSL
OFDM is used in ADSL connections that follow the
G.DMT (ITU G.992.1) standard, in which existing copper
wires are used to achieve high-speed data connections.
 COFDM vs VSB
• The question of the relative technical merits of COFDM versus
8VSB has been a subject of some controversy, especially
between Europe and USA. The United States has rejected
several proposals to adopt COFDM for its digital television
services, and has instead opted for 8VSB (
vestigial sideband modulation) operation.
• One of the major benefits provided by COFDM is that it renders
radio broadcasts relatively immune to multipath distortion and
signal fading due to atmospheric conditions or passing aircraft.
Proponents of COFDM argue that it resists multipath far better
than 8VSB. Early 8VSB DTV (digital television) receivers often
had difficulty receiving a signal in urban environments.
• However, newer 8VSB receivers are far better at dealing with
multipath, hence the difference in performance may diminish
with advances in demodulator design. Moreover, 8VSB
modulation requires less power to transmit a signal the same
distance, i.e., the received carrier-to-noise threshold is lower for
the same bit error rate. In less-populated areas, 8VSB may have
an advantage because of this. In urban areas, however, COFDM
is believed to offer better reception than 8VSB.
 Digital radio
• COFDM is also used for other radio standards, for
digital audio broadcasting (DAB), the standard for
digital audio broadcasting at VHF frequencies, and
also for Digital Radio Mondiale (DRM), the standard
for digital broadcasting at shortwave and mediumwave
frequencies (below 30 MHz).
• The USA again uses an alternate standard, a
proprietary system developed by iBiquity dubbed "
HD Radio". However, it uses COFDM as the
underlying broadcast technology to add digital audio to
AM (medium wave) and FM broadcasts.
• Both Digital Radio Mondiale and HD Radio are
classified as IBOC (in-band on-channel) systems,
unlike Eureka 147 (DAB: Digital audio broadcasting)
which uses separate VHF or UHF frequency bands
instead.
Digital Radio Mondiale (DRM)
• Digital Radio Mondiale (DRM) is a set of
digital audio broadcasting technologies designed
to work over the bands currently used for AM
broadcast, particularly shortwave. DRM can fit
more channels than AM, at higher quality, into a
given amount of bandwidth, using various
MPEG-4 codecs.
• It is also the name of the international non-profit
consortium designing and implementing the
platform.
 DRM Features
• DRM can deliver FM-comparable sound quality, but on
frequencies below 30 MHz (long wave, medium wave and
short wave), which allow for very-long-distance signal
propagation. VHF is also under consideration. DRM has been
designed especially to use older AM transmitters, avoiding
major new investments. DRM is robust against the fading and
interference which often plagues conventional broadcasting on
these frequency ranges.
• The encoding and decoding can be performed with
digital signal processing, so that a cheap embedded computer
with a conventional transmitter and receiver can perform the
rather complex encoding and decoding.
• DRM can transmit other data besides the audio channels (
datacasting) — as well as RDS-type metadata or
program-associated data as Digital Audio Broadcasting (DAB)
does.
• Unlike most other DAB systems, DRM uses
in-band on-channel technology and can operate in a hybrid
mode called Single Channel Simulcast, simulcasting both
analog signal and digital signal.
 DRM Modulation
• The modulation used for DRM is COFDM
(Coded Orthogonal Frequency Division
Multiplexing), where every carrier is modulated
with QAM (Quadrature Amplitude Modulation)
with a choose able error coding.

DRM Test
• DRM has been tested successfully on shortwave
, medium wave (with 9 as well as 10 kHz
channel spacing) and long wave.
 DRM Plus
• While DRM currently covers the broadcasting bands
below 30 MHz, the DRM consortium voted in March 2005
to begin the process of extending the system to the VHF
bands up to 120 MHz. DRM Plus (DRM+) will be the
name of this technology. Design, development and
testing are expected to be completed by 2007-2009.
• Wider bandwidth channels will be used, which will allow
radio stations to use higher bit rates, thus providing
higher audio quality. One likely channel bandwidth is 50
kHz, which will allow DRM Plus to carry radio stations at
near CD-quality. A 100 kHz DRM+ channel has sufficient
capacity to carry one mobile TV channel: it would be
feasible to distribute mobile TV over DRM+ rather than
DMB or DVB-H.