Digital Audio Broadcasting

1.0 INTRODUCTION

At present Digital Audio Broadcasting (DAB) is an European standard in the digital broadcasting system developed by the European Community Eureka 147 project, prompted by the EBU (European broadcasting Union). The objective was to specify a valid digital broadcasting system for terrestrial and satellite communications that would mitigate the odds of analogue broadcasting.

The use of AM and FM analogue standards have been widely used and have made substantial impact in the Broadcast industry for many years. However broadcasting with analogue technology suffers signal degradation problems, accumulating noise and distortions during each one of the phases that it goes through. These issues lies in the fact that the signal is not digital adding to that is the flaws that are prevalent in its modulation and systems design not forgetting frequency shortages but on the other hand, with digital technology, the signal suffers less degradation, since error correction methods are included to correct distortions that can alter the information thus providing high-quality sound. Thus, digital information is easily transportable and storable, also using up less space (spectral efficiency). DAB uses COFDM, a multicarrier that spreads data creating robustness against frequency selectivity, supporting multi-programme digital sound and data broadcasting services - not only for reception by fixed receivers but particularly for in-car and portable reception; this is made possible because of the modulation techniques employed.

[1] EBU Technical Review Autumn 1995 Kozamernik.

Eureka 147- DAB

The Eureka 147 Digital Audio Broadcasting standard was developed by a consortium of European organisations in 1987 with the aim to deliver a system that will satisfy specific requirements for digital audio broadcasting. It was known as the Eureka 147 project - DAB. Membership was formerly restricted but has since been extended to include various organisations, broadcast operators and other interested parties. Currently Eureka 147 has been merged with WorldDMB forum and manages all DAB family standards.

Technical Overview of DAB - Concepts

The basic and key technical concepts of DAB include the use of specific techniques for audio coding, modulation and transmission. COFDM, is one key technology used in DAB, which enables robust error correction, multiplexing and orthogonal features. Also mentioned are DPQSK modulation scheme and other important concepts in DAB.

Audio Coding in DAB

DATA REDUCTION

A stereo DAB signal might occupy 2 MHz of bandwidth, while an analogue FM broadcast requires 200 kHz. Thus, data reduction methods have been applied in DAB because of the excessive bandwidth requirements. DAB uses an existing audio compression technology, called MPEG Audio Layer 2 (also known as Musicam), which provides a capacity reduction factor of 7, needing only 192 Kbit/s [ ].

In the compression process the audio signal is first split by a filterbank (a poly-phase quadrature mirror filter (PQMF)) into 32 equal-bandwidth 'sub-bands'. MPEG Audio Layer 2 system analysis the sound ( by human ear psychoacoustic characteristics) and determines which of the 32 sub-bands will be perceptible by humans in the presence of weak frequencies and which won't be the. This way, the frequencies of the 32 sub-bands that are irrelevant or redundant to the ear are reduced by elimination and the rest encoded.

FREQUENCY RESOLUTION FOR MP2

Referring to Nyquist's Sampling Theorem:

Fs >= 2 B

where Fs is the sampling frequency and B is the bandwidth of the signal. For a sampling frequency of 48 kHz, which the DAB system uses, MP2's frequency resolution (the bandwidth of each subband) is:

MP2 frequency resolution = (48 kHz / 2) / 32 = 750Hz

This compression capacity aids to fit up to 6 stereo music programs in the bandwidth of just one channel (1.5 MHz) providing DAB with an ability not available in traditional analogue transmission. Also, this capacity is flexible so that, if stereo quality is not needed in a program, the space can be divided to fit, for example, two monophonic quality programs.

MODULATION

COFDM

FREQUENCY-DIVISION MULTIPLEXING

Priority is given to minimizing the interference or crosstalk among the channels and symbols comprising the data stream therefore instead of a having a single digitally modulated carrier with a very high symbol rate rather the bit-stream is sent through a time interleaver before it is multiplexed with the other programmes to form an ensemble. This means that it will be possible to increase the number of radio stations initially by a factor of at least three when compared with FM without congesting the radio waves. The ensemble bit-stream is fragmented into 1536 individual OFDM symbols or subcarriers at 1KHz spacing [ Vert] ; each sub carrier carries part of the entire transmitted data stream at a very low rate. Due to the low data rate of each RF carrier, any delayed reflections of the signal (i.e. “passive echoes”) add in a constructive manner to the direct signal already received.

By this majority of the data will be without distortion while the distorted single sub carrier's data can easily be repaired by the error correction algorithm which makes the system more complex as compared to the FM/AM transmissions which are simpler and not as rugged for high data transmissions.

The ability to multiplex gives room for data, multimedia as well as audio bits.

ORTHOGONALITY

The arrangement of the carrier in an FDM signal so that the sidebands of the individual carriers overlap maintaining the space between the nulls as Δf = 1/Δt. Explicit filtering process could be avoided. COFDM clearly specifies that the carriers are evenly spaced by precisely Δf = 1/Δt, where Δf = carrier spacing and Δt = OFDM symbol duration; so majority of the signals can be received without adjacent carrier interference.

CODING

DAB from its design can be seen as a highly flexible and dynamically reconfigurable system. It can accommodate a range of bit rates between 8 and 384 kbit/s(3) thanks to the use of Orthongonal frequency multiplexing but this alone is not sufficient especially for mobile radio, a range of channel protection mechanisms need to be put in place thereby increasing the length of the OFDM period this is a key attribute that is leveraged as shown in the below.[1]

Depending on the nature and the sensitivity of data specific choices of error correcting and convolutional coding can be adopted, but in the case of audio data, DAB adopts a technique that uses less space on air by providing more protection to more important bits in the audio frame together with a cyclic redundancy check

An example showing how more protection has been given to 128kbps of audio data is shown below

REED-SOLOMON CODE

In the case of the Reed-Solomon error protection, the data are assembled into packets of a certain length and these are provided with a special checksum of a particular length. This checksum allows not only errors to be detected but also a certain number of errors to be corrected. The number of errors which can be corrected is a direct function of the length of the checksum.

WHY FFT in COFDM

In COFDM or OFDM where many carriers are used these carriers are defined in the frequency domain and each as one element of a discrete Fourier spectrum. When data has to be transmitted to the receiver frequency domain defined carriers are transformed using an inverse FFT into the time domain, at the receiver the reverse process is applied to recover the data. This is accomplished by the VLSI (responsible for carrier orthogonality) at the transmitter (IFFT) and a corresponding (FFT) at the receiver.

DIFFERENTIAL QUADRATURE-PHASE-SHIFT-KEYING (DQPSK)

According to the standard pattern of DAB system, the QPSK symbols are created by using interleaving process in the block of bits before mapping them onto symbols, Frequency interleaving consists of a rearrangement of the symbol data stream over the carriers; For example, when a part of the channel bandwidth is faded, frequency interleaving ensures that the bit errors that would result from those subcarriers in the faded part of the bandwidth are spread out in the bit-stream rather than being concentrated. Frequency interleaving is present in a DAB system in order to eliminate the effects of selective fading prominent in AM broadcast where the whole channel bandwidth is faded at the same time. Frequency interleaving offers little to no benefit for narrowband channels that suffers from flat-fading (where the whole channel bandwidth is faded at the same time).

For QPSK modulation the mapping table is the rule according to which the data stream data(t) is converted to modulation signals i(t) and q(t).

There is no limit on what modulation to be used it all depends on where it will be applied but for the Eureka DAB the data stream data(t) of each sub-carrier is converted to the two modulation signals i(t) for the I path and q(t) for the Q path by means of a mapper (modulated using p/4-shift differential QPSK). In the case of p/4 DQPSK, pairs of bits ( b1 n and b2n in the mapping table) are combined to form a dibit. For dibit combination 10, for example, the mapper outputs the signals i(t)= -1 V and q(t)= -1 V according to the mapping table shown here.

Bit 1

Bit 0

DYk

i path

Q path

0

0

p/4

+1

+1

0

1

3p/4

-1

+1

1

0

-p/4

-1

-1

1

1

-3p/4

+1

-1

In this case the i(t) and q(t) have half the data rate of data(t) relative to the input rate and in turn modulates the carrier signal switching it only in its phase when compared to simpler transmission the available bandwidth capacity is now boosted by a factor of 2. The signal constellation results to eight points, but at any instant in time transmission is limited to a subset of four points, the x's or the o's. the benefit of this is that the filtered envelope of p/4 DPQSK does not pass through the origin which mitigates the effects of crossover distortion in the transmitter amplifier [h] sandeep chennakeshu and grey J. Saulnier, “Differential Detection of a pi/4 shifted DQPSK for Digital, 1999

In this particular case the phase of a signal is compared to the phase of the previous signal and thereof the original input is determined saving channel correction facilities hence the receiver becomes simpler thereby yielding the lowest BER for a given signal strength, but there is 3dB performance lost compared with coherent modulation. The Viterbi softer decoding is used at the receiver to lead to high coding gain. [z]

The quadrature amplitude modulations (QAM), like QPSK, 16QAM, 64QAM, etc, have been proposed as modulation schemes for DVB-T system [6]. Hence, the number of information bits in a modulated symbol can be increased in order to improve the spectrum efficiency.

GUARD INTERVAL

Due to the way the COFDM receiver locks onto the COFDM symbols a considerable payload is added to the data stream which accommodates the guard interval thereby producing longer symbols than (single - carrier transmission) reducing impairment form echoes provided that the echo has a delay less than the guard interval. The guard interval is included in between the processed symbols they are responsible to decay echoes thus preventing inter-symbol interference

Cellular Radio”, IEEE Trans. On Vehicular Technology, Vol. 42, no 1, pp.46 -57, Feb. 1993.

SYNCHRONISATION

With a robust and rugged modulation scheme employed if the wrong set of samples are processed the Fourier transform will not correctly decode the data on the carriers at the receiver, DAB overcomes this issues with synchronisation at the receiver by detecting and generating the null symbol for approximate timing by an analogue sync pulse.

In situations where the signals are not time repetitive therefore does not follow Δf = 1/Δt as required for the fast Fourier transform to be applied, more than one complete set of time samples are transmitted, which in transform theorem is a continuous repetitive sequence. The guard interval serves as an additional complete symbol therefore the initial timing accuracy only has to make sure that the samples to be subject to the FFT are gotten from one symbol. The longer the guard interval the more rugged the system, but a compromise has to be made for power to transmit the guard interval.

In DAB in order to improve synchronisation, a much more accurate timing is obtained if there is a correlation between the reference symbol (signal transmitted after the null symbol) with the known transmitted signal providing an impulse response of the channel.

NEW AND ONGOING DEVELOPMENTS IN DAB

The characteristic of DAB makes it capable of delivering reliable CD quality mobile reception [2]. Previous experiment in [3] shows that the DAB system with MPEG-1 video data compression scheme is not able to provide a good image quality over a mobile environment. Improvement in MPEG-1 has lead to MPEG-2 and MPEG-4 algorithms which can be used to increase the bit rates (about 1Mbits/s), therefore high-speed multimedia can be broadcast by DAB system this is known as Digital Multimedia Broadcasting (DMB), which is one of the applications that have emerged form the Eureka-147 DAB system [3, 4]. The main framework in DAB system is adopted by DMB system to process multimedia contents. Furthermore, the Reed-Solomon code is also adopted to assure that the bit error rate (BER) of less than 10-10 can be realized [5].

T-MMB supports DQPSK, 8DPSK and 16DAPSK. This results in high spectrum efficiency under complex multipath and time varying environment: when 384kbps per program, it can accommodate 4.5 programs (8DPSK, 1.5MHz) or 22.5 programs (8DPSK, 8MHz), or 6 programs (16DAPSK, 1.5MHz) or 30 programs (16DAPSK, 8MHz)

To counteract performance punishment from higher order constellation used in the T-MMB (Terrestrial Mobile Multimedia Broadcasting) standard, T-MMB employs a well-designed LDPC and a new scrambler to support it. The well-designed low density parity check (LDPC) specifies a short code (length 4608) which can reduce storing requirements and features as most rapid convergence with noise floor smaller than 10-9

By employing one or more effective PAPR reduction technique it's been possible to upgrade Harris DAB660 from DAB/T-DMB to T-MMB. Nufront Tech., 2009

5.0 CRITICAL APPRAISAL

SPECTRAL SPACE

It is true to say that DAB systems cohabits spectral space with existing applications. On the other hand shared-spectrum techniques could be used to locate the digital signal in the FM and AM bands. But regardless of any future DAB implementation, FM and AM broadcasts ultimately determine how the spectral space is redistributed.

COMPACTIBILITY WITH OLDER SYSTEMS

The DAB system chain includes several blocks which introduce a significant processing delay. For example, the time interleaver introduces a delay of 384 ms, and the audio coder/decoder introduces a delay of several tens of milliseconds. The total delay in the system may vary from one implementation to the next. The system delay should be taken into account when the receiver switches between DAB and FM “simulcast” programmes, so that a seamless transition is obtained. It will become necessary for simulcast FM transmissions to be delayed by nominally the same amount, say one second, regardless of the receiver design. This nominal delay should be taken into account when signalling the current time information.

DAB+

An ongoing development in DAB is targeted to be able to increase the capacity of the system to broadcast more supported by more rugged and efficient audio coding. MP2 has had issues with its 750 Hz frequency resolution, which stems from the fact that the input signals were only divided into 32 subbands.The frequency resolution determines is how finely redundancy can be removed (key attribute to reduced bit rate audio is coding). With MP2's 32 subbands, if there is a frequency component that was deemed to be perceptible in a subband then that whole subband must be encoded. In comparison, AAC offers a better frequency resolution of 23Hz which allows just those frequency components that are perceptible to be encoded, and the rest discarded. Redundancy is cut down more.

Overall, MP2 should not really ever be used at bit rates below 192 kbps, and 128 kbps is simply far too low a bit rate to provide audio quality that should be expected on a modern digital radio system.

AAC+ is the combination of the standard AAC (LC AAC -- low complexity AAC) codec with Coding Technologies' Spectral Band Replication (SBR) technology -- the AAC audio codec encodes the bottom half of the audio spectrum, and SBR encodes the top half of the audio spectrum, due to AAC+'s use of SBR, which only consumes a bit rate of between 1 to 3 kbps to encode the entire top half of the audio spectrum, compared to MP2 that consumes a relatively large (not 50%, but still a very sizeable percentage) of the overall bit rate on encoding the upper half of the spectrum.

Thus greater spectral efficiency is offered by the DAB+ system, because a stronger error correction coding scheme can correct more errors than a weaker one, so stronger error correction coding schemes enable the capacity of a multiplex to increase.

CONCLUSION

The characteristics of Digital audio broadcasting has opened up great potential for many reasons: governments are facing the hard task of sharing out the finite radio spectrum between a mass of conflicting interests and they welcome DAB as a highly spectrum-efficient system; broadcasters see the opportunity to offer more services of better quality and presentation and make more money; manufacturers welcome the opportunity to sell large quantities of DAB receivers and associated equipment, and network operators are keen to build the new distribution and transmitter networks that are required for DAB terrestrial services. Not least, the listener welcomes a new technology which offers more choice and higher technical quality, as well as a very robust signal even on the move.

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