analogue radio vs

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Analogue Radio vs. Digital Radio

Traditional radio systems such as AM/FM/LW/SW systems have always used analoguetechnology. DAB is the first (to my knowledge) major digital system designed specificallyfor digital broadcast radio.

Analogue Radio

Analogue radio transmission consists of transmitting the actual audio signal modulated ontothe RF carrier. Analogue basically means that the signal can take on any value (within thelimits set by the transmitter). The problem with transmitting analogue audio signals is that

because any noise, interference or self-interference (multipath effect) is added to the signal atany point then this cannot be removed from the audio signal and this degrades the audioquality of the signal or causes hiss.

Digital Radio

Digital radio systems such as DAB or digital radio that is delivered via digital satellite (DSat)or Freeview, addresses the disadvantages that hamper the analogue transmission systems(although these can usually be overcome by improving your FM reception by purchasing a

better FM aerial and/or relocating the FM aerial). Digital radio systems consist oftransmitting digital waveforms on the carrier to the receiver. These waveforms are thendecoded to binary format to make up the digital words that carry the amplitude values of theaudio waveform (this is expanded upon on the MPEG Coding page).

At the radio receiver, the carrier part of the signal is removed by downconverting fromradio frequency (RF) to low frequency (termed 'baseband') which just leaves the digital signalalong with the noise and interference that has been added to the signal at the transmitter, in

the air, and at the receiver itself. The receiver then decides which symbols were transmitted.In the simple binary case, the digital 0s and 1s will be transmitted as equal amplitude butopposite in sign. The receiver will then just look at whether the signal is above or below zerovolts and for example if the voltage is above zero volts then it will decide the bit transmittedwas a 1 and if it is below zero volts then it will decide that a 0 was transmitted. Because ofthe noise and interference the voltage value at the decision instant might not be the same signas the bit that was transmitted and a bit error will be made. The proportion of the bits thatare received in error (the bit error rate, or BER, calculated by dividing the total number of biterrors by the total number of bits transmitted) is directly related to the received signal power,which is why installing an external aerial pays off because an external aerial will invariablyreceive a higher signal power than an internal aerial. Unfortunately there are strict limitations

to the power level at which broadcasters are allowed to transmit at. This is a necessarylimitation because otherwise the transmitted signals would cause too much interference inadjacent frequency bands.

DAB and digital terrestrial TV (DTT) use a more advanced form of digital modulation than isexplained above. Both of these systems use COFDM, which stands for Coded OrthogonalFrequency Multiplexing.

Error Correction

The main advantage of transmitting radio digitally is that the bit errors that are made when

the receiver chooses the incorrect symbol can in most cases be corrected. This is achieved byusing forward error correction coding (FEC coding). The FEC encoder adds redundant bits to
http://www.digitalradiotech.co.uk/mpeg_coding.htmhttp://www.digitalradiotech.co.uk/cofdm.htmhttp://www.digitalradiotech.co.uk/cofdm.htmhttp://www.digitalradiotech.co.uk/mpeg_coding.htm


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the original bitstream so that the receiver can decode the signal and in the vast majority ofcases correct any errors. An example of a very simple error correction code (this is not a

practical code because it is very inefficient and is not very effective at correcting errors) usesthe rule that if a 1 is to be transmitted then it is repeated 3 times. Then the receiver takes amajority decision so that it decides that a 1 was transmitted if 2 or 3 ones are received.

Therefore if one error is made out of the three bits transmitted that error can be corrected.This is an example of a block error correction code because a group of input bits (in this caseone input bit) are transmitted as a larger block of coded bits. A measure of the redundancyand of the power of an error correction code is its code rate which is given by the number ofinput bits to the forward error correction encoder at the transmitter divided by the number oftransmitted bits. For the example above, one input bit goes into the FEC encoder at a timeand three come out. Therefore its coding rate equals 1/3. The lower the value of the codingrate the more powerful the code will be at correcting errors and vice versa. For wirelesstransmission, block coding is not the primary form of FEC coding, although it may be used aswell. The preferred type of FEC coding for wireless systems is called convolutional codingand is more powerful than block coding. Typical code rate values used are 1/3, 1/2, 2/3 and

3/4.The type of FEC coding used on DAB is a form of convolutional coding where different partsof the audio bitstream use different code rates. For example, scale factors are shared betweena lot of samples and therefore it is more important that these scale factors should not bereceived in error compared to the individual samples, so a lower code rate (higher protectionlevel) is used to protect the scale factors than the individual samples.

A similar variable code rate will be applied to the video stream on DTT.

Introduction to Digital Audio Broadcasting (DAB)

The DAB system was designed in the late 1980s, and its main original objectives were toprovide radio at CD-quality; to provide better in-car reception quality than on FM; to use thespectrum more efficiently; to allow tuning by the name of the station rather than byfrequency; and to allow data to be transmitted. DAB fulfills most of these objectives, but withone rather important exception: DAB sounds worse than FM.

Why is the sound quality so bad?

The main reason why there is a problem with the audio quality on DAB is due to the

broadcasters using bit rate levels that are too low to provide good audio quality. The reasonwhy they're using insufficient bit rate levels is due to DAB using the inefficient MP2 audiocodec, which needs to be used at bit rate levels of at least 192 kbps to provide good audioquality -- FM provides an audio quality which is equivalent to 192 - 224 kbps MP2.Unfortunately, 98% of all of the stereo stations on DAB in the UK are using a bit rate level of128 kbps, hence the audio quality is poor. This problem of using low bit rates doesn't onlyaffect the UK, either, because the handful of other countries that are trying to promote the oldDAB system -- Denmark, Norway and Switzerland -- are also using low bit rate levels.

The reason why such low bit rate levels are being used is because the broadcasters havedecided to launch quite a lot of new digital-only stations, but as there is only a limited amountof spectrum available for DAB to use, the broadcasters decided to use low bit rate levels in

order to fit these new stations onto DAB even though they knew full well that the audioquality would be lower than on FM.


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The broadcasters and Ofcom try to make this out as being a "trade-off", but the reality is thataudio quality was sacrificed in order to provide more stations. For example, the broadcastersdecided to use 128 kbps for stereo stations, and this allows 9 stations to be carried in a DABmultiplex. If they reduced the number of stereo stations to 8 rather than 9 then half of thestations could transmit at a bit rate of 160 kbps, which would provide a significant

improvement in quality, albeit that it would still sound worse than on FM.

The Incompetent Adoption of DAB in the UK

Justification for the use of the word "incompetent"

When you look at the history of radio broadcasting, one of the most striking things is howlong the systems have lasted:

AM radio was first commercially broadcast in 1920

FM was invented in 1935, there was an FM broadcast band in the US in the

1940s, and the Zenith-GE pilot tone system was standardised in 1961 toprovide FM stereo, and FM stereo has remained unchanged up to thepresent day

DAB on the other hand was "properly" launched in the UK in 2002, yet just 3 years later theWorldDAB Forum pulled the plug on the old DAB system by ordering that the AAC+ audiocodec be adopted, which led to the design of the new DAB+ system, which will make allDAB receivers obsolete in the coming years.

Because 3 years is such an extremely short duration in broadcasting system terms, thelaunch of the old DAB system in the UK has got to go down as the most incompetenttechnical decision ever made in the history of broadcasting -- and that includes both TVand radio.

"Incompetent" is a strong word to use, but I'm afraid that in this instance I feel it is perfectlyjusified.

And if you're now thinking "it's easy to say this in hindsight", the technologies had existed foryears before DAB was launched in the UK, but the BBC didn't upgrade DAB prior tolaunching it. For example, the AAC audio codec had been standardised in 1997, and Reed-Solomon error correction coding is used as the error correction on CDs, so it had been inwidespread use since the 1980s. If these two technologies had been used to upgrade the DABsystem prior to it being launched, the audio quality and the reception quality would be far

better than they are with the current system, and DAB would actually be able to carry all ofthe analogue stations, whereas Ofcom has admitted that around 90 analogue stations will

never be able to fit on DAB due to either being unable to afford the sky-high transmissioncosts or due to the local DAB multiplexes being full.

Of course there have been other broadcasting system failures, but nothing comes close tomatching the significance of the launch of a system that was meant to be the digitalreplacement for the ubiquitous FM system followed just 3 years later by its ruling bodyscrapping it.

It also should not be forgotten that when the UK launched DAB in 2002 they did so thinkingthey were pioneering DAB and that the rest of Europe and then most of the rest of the worldwould naturally follow their lead. But DAB+ was only actually designed because this planwent so incredibly wrong that the only countries the UK could get to commit to using the oldDAB system were Denmark and then more recently Norway, with virtually every othercountry that said anything on the subject of digital radio being opposed to using the old DABsystem.


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Put simply, if DAB+ hadn't been designed, the UK, Denmark and Norway would have beenthe only countries stuck using the old DAB system, while all other countries would haveadopted one of the modern and far more efficient systems that can be used to carry digitalradio, such as DVB-H, T-DMB, HD Radio or DRM+. And the saddest thing about this wholestory is that it was so easily avoidable, for the reasons I will expand upon below.

The "true" launch of DAB in 2002

Although the BBC began transmitting DAB in 1995, there were no DAB receivers on sale atall until Arcam brought out its Alpha 10 tuner in December 1999, which cost 800.

But DAB was only "properly" launched in March 2002 when the BBC launched 6 Music,which was the first of five new BBC digital-only stations to launch that year. But the crucialelement that made 2002 the true launch-date of DAB was the beginning of the advertising

blitz that coincided with the launch of the BBC's digital-only stations, and as of today(18/4/07) the BBC has broadcast 19 high-impact TV advertising campaigns for DAB, which

I've calculated would have cost 155 million

1

if the BBC had to pay for these adverts to bebroadcast on commercial TV -- with DAB receiver sales standing at around the 4 millionmark, that means the BBC has pseudo-subsidised DAB to the tune of 155m / 4m = 38.75

per DAB radio!

Then just three and a half years later in October 2005, the WorldDAB Forum (now called theWorldDMB Forum) ordered the Technical Committee to do the work necessary to add theAAC+ audio codec, which was a decision that will make all of the existing DAB receiversobsolete in the coming years.

The design of the old DAB system

This section is a short summary of how the old DAB system was designed, but for a slightlymore in-depth description see here.

The old DAB system was designed in the late 1980s and early 1990s, and all of the maincomponent technologies -- the audio codec, the modulation and error correction coding -- had

been chosen by early 1991, and they remain unchanged to the present day.

The designers of the old DAB system chose to use low complexity technologies, which wasmainly due to the fact that microprocessors weren't very powerful in the late 1980s. Examplesof this were that the designers chose to use the MP2 audio codec instead of MP3, and theychose to use simple rather than the more complex but stronger error correction coding theycould have used.

The downside of choosing to use these low complexity technologies was that they made DABan incredibly inefficient system. For example, whereas MP3 was designed to be used at 128kbps, MP2 was designed to be used at bit rate levels between 192 - 256 kbps, and this is

borne out by the following quote in a BBC R&D report about DAB from 1994:

"A value of 256 kbit/s has been judged to provide ahigh quality stereo broadcast signal. However, a smallreduction, to 224 kbit/s is often adequate, and in somecases it may be possible to accept a further reduction to192 kbit/s, especially if redundancy in the stereo signalis exploited by a process of 'joint stereo' encoding (i.e.some sounds appearing at the centre of the stereo image
http://www.digitalradiotech.co.uk/design_of_dab.htmhttp://www.bbc.co.uk/rd/pubs/whp/whp-pdf-files/WHP061.pdfhttp://www.digitalradiotech.co.uk/design_of_dab.htmhttp://www.bbc.co.uk/rd/pubs/whp/whp-pdf-files/WHP061.pdf


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need not be sent twice). At 192 kbit/s, it is relativelyeasy to hear imperfections in critical audio material."

And combining the inefficiency of the MP2 audio codec with the weak error correctioncoding used on DAB (weak error correction coding leads to DAB multiplexes having a lowdata capacity -- i.e. a low spectral efficiency), DAB multiplexes can only carry a very smallnumber of radio stations:

Stereo radio

station bit rate

kbps

Audio quality level Numbe

r of

station

s per

DAB

multipl

ex

Bandwi

dth per

station1

kHz

256 Near CD-quality 4 428

224 FM-quality 5 342

192 Near FM-quality 6 285

1 - the bandwidth of a DAB multiplex is 1,710 kHz

DAB at the BBC -- 1990 - 2002The following bullet points are a time-line of some relevant events to do with DAB at theBBC in the 1990s and early 2000s:

January 1990 - BBC R&D department first trialed DAB by transmitting itfrom Crystal Palace

1991 - BBC R&D demonstrated DAB to press

1992 - "Extending Choice: The BBC's Role in the New Broadcasting Age"document was published

September 1995 - BBC national DAB multiplex began broadcasting Radios1-5 nationally in 1995 (BBC World Service was added at a later date)

May 1996 - "Extending Choice in the Digital Age" document published byBBC Director-General John Birt

September 1998 - bbc.co.uk mentions BBC's plans to launch 4 digital-onlyradio stations (BBC Paliament, Asian Network, new music station, newsports station)

December 1999 - The first DAB tuner went on sale -- the Arcam Alpha 10tuner -- costing 800

Summer 2001 - BBC deliberately withholds information that the launch ofnew stations will drastically degrade the audio quality of existing stationsin its public consultation for new digital radio stations (this was admittedin an email in 2002 by the then Controller of Radio & Music Interactive,Simon Nelson)
http://news.bbc.co.uk/1/hi/entertainment/174535.stmhttp://news.bbc.co.uk/1/hi/entertainment/174535.stmhttp://news.bbc.co.uk/1/hi/entertainment/174535.stmhttp://news.bbc.co.uk/1/hi/entertainment/174535.stm


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November 2001 - Mediocre feedback from consultation (2 of the proposedstations get less than 50% acceptance figures and the only station to getmore than 60% acceptance was BBC7)

2002 - 5 new digital-only stations are launched (6 Music, 1Xtra, BBC7,Asian Network, Radio 5 Sports Extra)

Soon after the BBC's national DAB multiplex was launched in 1995, it was carrying thefollowing stations:

Station Bit rate

kbps

Radio 1 192

Radio 2 192

Radio 3 192

Radio 4 192

Radio 5 96

World

Service96

Space left

over192

The BBC will have known from the early 1990s -- straight after the listening tests at SwedishRadio in 1990 that led to the MP2 audio codec being adopted for DAB -- that DAB wouldneed to use a bit rate level of 256 kbps to provide near CD-quality (the provision of near CD-quality was the main reason why DAB was designed in the first place), and yet when theBBC launched its national DAB multiplex in 1995 it obviously couldn't use a bit rate level of256 kbps or else Radio 5 and the World Service wouldn't have been able to be carried on themultiplex. So, instead, the BBC reduced the bit rate levels of Radios 1-4 to 192 kbps -- i.e.when they launched their national DAB multiplex they had already reduced the audio quality

level to below the 224 kbps require to match FM! However, this didn't really matter at thetime too much, because you couldn't buy a DAB receiver until December 1999, and then theycost 800.

It is just staggeringly incompetent that the BBC knew from the early 1990s that theywould be adding a number of new digital-only stations to their national DAB multiplex,yet they only had enough space left for one additional stereo station, and then in 2002they actually added 5 new digital-only stations. How long did they need to figure outthat DAB wasn't up to the job??

The bit rate levels of all of the BBC's music stations apart from Radio 3 are now half the bitrate levels that were suggested to be used originally and the audio quality is awful.


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Number of radio stations available

The majority of people in the UK can currently receive 4 DAB multiplexes, which consists ofthe BBC and Digital One national multiplexes, a regional and a local multiplex. Once theforthcoming DAB expansion has finished, the majority of people will be able to receive 5DAB multiplexes due to the addition of a second commercial national multiplex that will

launch in 2008. This means that if the recommended bit rate levels of 256 or 224 kbps hadbeen used to provide the near CD-quality or at least FM-quality that DAB was originallydesigned to provide, people would be able to receive the following number of stations:

Bit rate

kbps

Number of

stations available

up to 20081

Numb

er of

statio

ns

availa

ble

from

20081

256 20 25

224 24 30

1 - the table includes the effect of mono stations using half the bit rate of stereo stations -- 80% of radio stations on DAB are stereo stations and 20%are mono, and this works out as adding one extra station per multiplex whether 256 kbps or 224 kbps is used

The new spectrum for the expansion of DAB is all there's going to be DAB, because this newspectrum was acquired from the Regional Radio Conference (RRC-06) in Geneva last year,and the last time there was a frequency planning conference on that scale was in Stockholmin 1961!

Twenty-five to thirty radio stations -- or 36 stations tops in London -- was never goingto be enough, so the above table basically shows that it was inevitable that poor audioquality would be provided on DAB. I would therefore suggest that the BBC and theRadio Authority (the regulators of commercial radio pre-Ofcom) were grosslyincompetent to use a digital radio system where it was inevitable that poor audio qualitywould be provided.

Transmission costs per radio station

I was provided with some actual DAB and FM transmission cost figures once by someone inthe DAB industry, which are contained in the following table along with how much it wouldcost to transmit at 224 kbps or 256 kbps based on the fact that DAB transmission costs are

pro rata with the number of capacity units (CU) consumed (capacity units are usually but notalways linearly proportional to the bit rate levels):

Transmission

type

Coverage area Transmis

sion costper


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annum

256 kbps DAB Local 192,000



FM Local 60,000

According to Ofcom, 50% of all existing analogue radio stations make no profit. I wouldtherefore suggest that the above table shows that using DAB makes providing poor audioquality inevitable, because commercial radio stations wouldn't be able to afford to pay thetransmission costs to provide good audio quality. The Radio Authority was thereforeincompetent to propose that DAB should be used as the digital radio system in the UK.

The above table also goes some way to explain why only around 45% of all existing analogueradio stations are on DAB. The big stations owned by the big commercial radio groups are onDAB -- the big commercial radio groups own the commercial DAB multiplexes -- but thesmall and medium-sized stations are not on DAB, because most simply cannot afford thetransmission costs. DAB is a great way for the big commercial radio groups to monopolisedigital radio by excluding access to their competition.

Ofcom has said that even after the expansion of DAB, 90 out of the existing 326 commercialradio stations won't be able to transmit on DAB either due to not being able to afford it(which probably accounts for most of them) or because the multiplexes will be full.

Better technologies were ready and waiting to be used

When I've criticised DAB in the past numerous people have said "oh, it's easy to see mistakeswith the aid of 20/20 hindsight" or words to that effect. The reality is that technologies weresitting there ready and waiting to be used, but those in charge of DAB stuck their heads in thesand, and the people at the BBC, the Radio Authority and in commercial radio were probablytoo oblivious to there even being a problem.

As I mentioned earlier, the problem with DAB is that it is an extremely inefficient system, sobelow I'll say which technologies existed that could have vastly increased the efficiency,which would have avoided the current problems altogether.

Reed-Solomon error correction coding -- invented in 1960

Reed-Solomon (RS) coding is used as the error correction coding on CDs, and its efficacy foruse in conjunction with OFDM modulation had been shown in an EBU Technical Reviewarticle (edition 224) by Alard and Lasalle in August 1987, so it was obviously around whenDAB was originally being designed, but they obviously either ignored it or deemed that itwas too computationally complex. I'm afraid that if they deemed it to be too computationallycomplex then they were trying to design a digital radio system before digital processing wasfast enough to handle it, and they should have waited until Moore's Law caught up.

RS coding could have increased the multiplex data capacity by approximately 40%. RScoding has now been adopted for DAB+.


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MP3

MP3 was the joint-winner with MP2 of the listening test at Swedish Radio in 1990 that led tothe designers of DAB adopting MP2, so MP3 obviously could have been adopted at any pointduring the 1990s.

MP2 was targeted at bit rate levels between 192 - 256 kbps, whereas MP3 was optimised foruse at 128 kbps. However, MP3 at 128 kbps would have provided similar audio quality to192 kbps MP2, so MP3 was 50% more efficient than MP2.

AAC

Development of AAC began in 1993 when it was shown that superior compressionperformance could be achieved by removing the requirement for codecs to be backwardlycompatible with MP2 and MP3. AAC was standardised in 1997.

If the designers of DAB had spotted that their system was far too inefficient they could haveworked to adopt AAC in parallel to the ongoing development work and incorporate AAC

soon after it had been standardised in 1997. And considering that the first DAB receiverdidn't go on sale until December 1999, AAC obviously could have been adopted for DAB.

AAC is twice as efficient as MP2 -- ironically, it was in listening tests carried out by BBCR&D on AAC that proved that AAC was twice as efficient as MP2: first for a multi-channeltest in 1996, then for a stereo test in 1998. To be fair to the BBC R&D engineers, they did sayin subsequent documents how good AAC was, so presumably the BBC management nothaving a clue about technology that was to blame for the glaring error.

Combination of more efficient audio and error correction coding

The following table shows the combined effect of using the more efficient audio and error

correction coding relative to that used on DAB:

Audio + error

correction coding

Efficiency relative to

DAB

MP3 + RS coding 2.1

AAC + RS coding 2.8

SBR - mp3Pro & AAC+

Spectral Band Replication (SBR) is the technology that makes AAC+ more efficient thanAAC, and some of you may remember a codec that was around a few years ago calledmp3Pro, which, like AAC+, consists of the addition of SBR, but this time to MP3. So ifeither MP3 or AAC had been adopted then it would have been relatively painless to add SBRin order to make DAB even more efficient.

Conclusion

If either of the above two options had been adopted instead of the technologies that the oldDAB system actually uses then there wouldn't have been a problem, because the broadcasterswouldn't have needed to provide low audio quality in order to provide the number of stationsthey are providing in the UK.
http://sound.media.mit.edu/mpeg4/audio/public/w1420.pdfhttp://sound.media.mit.edu/mpeg4/audio/public/w1420.pdfhttp://sound.media.mit.edu/mpeg4/audio/public/w2006.pdfhttp://www.digitalradiotech.co.uk/documents/BBC_R&D_AAC_1999_Open_Day.pdfhttp://www.digitalradiotech.co.uk/documents/BBC_R&D_AAC_1999_Open_Day.pdfhttp://sound.media.mit.edu/mpeg4/audio/public/w1420.pdfhttp://sound.media.mit.edu/mpeg4/audio/public/w1420.pdfhttp://sound.media.mit.edu/mpeg4/audio/public/w2006.pdfhttp://www.digitalradiotech.co.uk/documents/BBC_R&D_AAC_1999_Open_Day.pdfhttp://www.digitalradiotech.co.uk/documents/BBC_R&D_AAC_1999_Open_Day.pdf


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However, unfortunately the above two options weren't used, and the audio quality on DAB islow, and it will be low for the next few years. The problem with the audio quality will besorted out once DAB+ has been fully adopted -- there will be no reason to provide low audioquality then, albeit that I wouldn't put it past some of the most tight-first commercial radiogroups continuing to provide the same low audio quality as they do now, in particular GCap

Media and Emap, where they actually seem to be proud of providing as low audio quality aspossible.

But it will be several years before all of the legacy MP2 services have been switched off,albeit that we will see DAB+ stations launch within the next 3 years or so. But MP2 andAAC+ services will have to go through a period of being transmitted in parallel before thetime comes when incompetent Ofcom allows the MP2 services to be switched off, so thequality on DAB is -- amazingly -- going to get worse before it gets better.

We've already had 5 years of a sub-standard service, so the effect of the incompetent adoptionof DAB in the UK will have lasted a long time before the final MP2 service has beenswitched off.

COFDM

The modulation scheme that DAB uses is Coded Orthogonal Frequency DivisionMultiplexing (COFDM). COFDM uses a very different method of transmission to olderdigital radio modulation schemes and has been specifically designed to combat the effects ofmultipath interference for mobile receivers.

Multipath

is the term for the different paths that a signal takes in reaching an aerial from the transmitter.For example, one path may be a line-of-sight path from the transmitter to the aerial whereasanother path may bounce off a hill or building before reaching the aerial. In this example, thesignal that travels along the line-of-sight path arrives at the aerial first followed a short periodlater by the path that has bounced off the hill or building.

As the different paths travelled are of different length the time taken for the signal to reach

the receiver will be different, with the direct path (if there is one) reaching the receiver first,followed by reflected paths. The effect that these multipaths have on the received signal at the


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antenna is that the amplitude of the received signal fluctuates. The reason for this fluctuationis due to the relative phase angle between the different paths. The received signal is very-highfrequency sinusoidal carrier signal with comparatively a very slowly changing informationsignal that has been modulated onto the carrier. Therefore, a good way to model a carriersignal is to ignore the low frequency modulating signal and just assume that the multipaths

are each high frequency sinusoids with different amplitudes due to the different distancescovered (the amplitude reduces the further it travels) and relative phase angle due to thedifferent delay. To find out the instantaneous amplitude that is received at the antenna avector diagram can be drawn on which each multipath is represented by its amplitude (thelength of the vector) and its phase angle relative to, say, the phase angle of the direct path(which gives the vector's direction). An example of a vector diagram is given below (ignorethe N and E)

Ignoring the labels on the above diagram, the diagram could represent a two-path signalwhere the direct path is the pink vector and the sky blue vector is the delayed path, and thevector addition produces the red vector, and it is the resultant red vector that the receiveractually "sees".

As a mobile receiver moves relative to the transmitter the distances travelled by the paths alsochanges and because the wavelength of a radio signal is of the order of 3 metres for VHF FMsignals and about 1.5 metres for DAB signals in Band III the relative phase angles betweenthe paths changes rapidly and randomly. For example, if there were two multipaths that arein-phase (zero relative phase difference) then one of the paths only has to travel half awavelength further than the other (about 75 cm for Band III DAB signals) for the relative

phase of the path to change by 1800. If you look at the vector diagram above, if the bluevector had a relative phase of 1800 and a length equal to the pink vector then it would befacing in the opposite direction to the pink vector so the pink and sky blue vectors wouldcompletely cancel one another and the length of the resultant red vector would be zero. As Iexplained above, the antenna "sees" the red vector, so the amplitude that the antenna sees isalso zero. The term for this in physics is "destructive interference" and the signal is said to bein a "deep fade".

Deep fades occur more frequently the faster the mobile is travelling, but the duration that thesignal is in a deep fades decreases as the speed of the mobile increases. A typical graph of theamplitude of the carrier signal that the mobile antenna sees as it travels is shown below:


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Wideband & Narrowband Wireless Transmission

The effect of multipath fading in the frequency domain is that wideband signals suffer from"frequency selective fading", which means that different parts of the spectrum are faded morethan others. Narrowband signals on the other hand suffer from "flat fading" where the wholesignal spectrum fades, so for example, a narrowband signal's spectrum would be multiplied

by the above graph, which would mean that for example after travelling about 2.7 metres,destructive interference occurs and the whole spectrum will fade, hence the term 'flat fading'.

Whether a wireless digital communication system is wideband or narrowband depends on theduration of the transmitted symbols over the mobile channel. The mobile channel can berepresented by what is called a power delay profile, which shows the received power after the

transmission of a very short pulse, called an impulse, and the power of the signal receivedvaries with time due to the different multipaths that arrive at the receiver. The duration fromthe first received path to the last received path that has significant power gives the maximumdelay of the channel. A typical power delay profile varies between approximately 4 s forurban environments up to about 20 s for a rural environment.

A wireless digital communication system transmits "symbols" through the channel, forexample, for a single-carrier binary phase shift keying (BPSK, which uses either 00 phaseangle or 1800 phase angles, and a carrier phase of 00 represents a bit value of 0, and a carrier

phase of 1800 represents a bit value of 1, so each transmitted "symbol" represents one bit ofdata) modulation scheme then the symbol duration is the duration between when the phaseangles can change.

And a wireless digital communication system uses narrowband transmission if the channelsymbol duration is greater than the maximum delay of the mobile channel (e.g. 4 s for urbanand about 20 s for a rural environment) and the system is wideband otherwise.

In a digital wireless communication system, the bit errors are far more likely to occur whenthe signal is in a deep fade. Therefore these systems must mitigate the negative effects thatmultipath causes and different systems go about it in different ways. The two best knownmodern wireless digital communication transmission schemes are CDMA and OFDM.CDMA is used on the new 3G mobile phone system and is a wideband transmission scheme,which means that the channel symbols (which are called chips for CDMA) are far shorterthan the maximum delay of the mobile channel. OFDM, as used on DAB and Freeview

actually uses narrowband channels (subcarriers), but there are many of these narrowband


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channels transmitted in parallel, so the overall spectrum is wide (but this doesn't mean that ituses wideband transmission principles).

Error Correction Coding

The result of OFDM using a large number of narrowband subcarriers is that each subcarriersuffers from flat fading, as described above. Because the subcarriers are subject to flat fading,DAB uses COFDM (coded OFDM) which means that the data transmitted on the subcarriersis protected by forward error correction (FEC) coding. The type of error correction codingthat is used in COFDM is convolutional coding and the effect of convolutional coding is thatfor every one bit input to the error correction encoder, more than one bit is output dependingon the "code rate" being used. For example, a code rate of 1/3 would mean that for every bitinput to the error correction encoder, 3 bits will be output and these 3 bits are transmitted.Error correction coding therefore adds redundancy to the signal in order for the receiver to beable to correct any bits that are received in error. The error correction decoder used inCOFDM is the Viterbi algorithm which tries to decode what bits were sent depending on the

received sampled values.

COFDM also allows different groups of bits to be protected with a different strength coderate because some bits are more important for the correct reproduction of the audio than someof the other bits. For example, important parameters in the MPEG audio stream are the filter

parameters, so these are coded with a lower code rate (a lower code rate provides higherprotection as more redundancy is added) so that the Viterbi error correction decoder has ahigher chance of correcting any errors.

Interleaving

Unfortunately, the Viterbi algorithm performs poorly when it is presented with bit errors thatare all bunched together in the stream, and because the subcarriers are subject to flat fading

bit errors usually do occur in groups when a subcarrier is in a deep fade. To protect againstthis, DAB uses time interleaving and frequency interleaving.

An example of how time interleaving is used is shown in the above table. The data symbolsare written into the interleaving block in column order, then once the block is full, thesymbols are read out in row order, so for example the symbols would be read out in thefollowing order: 0, 8, 16, 24, 32, 1, 9, 17 and so on.

At the receiver, the received symbols are written into the same sized interleaving block in

row order, and once the block is full, the symbols are read out in column order to return thesymbols to the original order.


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The above mapping is called a Gray code mapping, because adjacent symbols (or in this casephase changes) only differ by the value of one bit, which lowers the probability of there beingtwo bit errors for one symbol.

After the 1536 pairs of bits have been mapped to one of the four phase changes these phasechanges are applied to the 1536 subcarriers. The previously transmitted QPSK symbol oneach subcarrier will be placed in memory in the transmitter, and the phase change will thenrotate this symbol. For example, if the previous transmitted symbol on a subcarrier was thetop, right-hand point (at 45o) in the figure below (called a signal constellation diagram) andthe bits that are being mapped onto the subcarrier are '11' then the phase will rotate by 180 o sothat the bottom, left-hand point (at 225o) will be transmitted on that subcarrier.

The QPSK symbols shown in the signal constellation diagram above are representednumerically by their co-ordinates on the diagram. The 'Re' axis is the 'real' axis and the 'Im'axis is the so-called 'imaginary' axis, which are the terms for diagrams that display what arecalled 'complex numbers'. A complex number consists of the combination of a real plus animaginary number:

I + j Q

where I is the real part of the complex number and Q is the imaginary part of the complex

number, and the 'j' is always multiplied by the imaginary number. The actual meaning of 'j' isthat it is equal to the square-root of -1, which doesn't actually exist, and that is why it is calledan imaginary number, but complex numbers are a very useful mathematical concept and thefact that the imaginary number doesn't actually exist doesn't matter.

To read an excellent tutorial about complex numbers and their use in digital signal processingdownload this Acrobat file: http://www.dspguru.com/info/tutor/QuadSignals.pdf(136 KB).

The transmitted symbols have the following (normalized) co-ordinates:

Rectangular Co- Carrier Phase
http://www.dspguru.com/info/tutor/QuadSignals.pdfhttp://www.dspguru.com/info/tutor/QuadSignals.pdf


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ordinate

0.707 + j0.707 45o

-0.707 + j0.707 135o

-0.707 - j0.707 225o

0.707 - j0.707 315o

Complex numbers are used to represent signal points on a constellation diagram because thereal and imaginary axes are at 900 apart and a sinewave and a cosine wave (both with thesame frequency) are also 900 out of phase. This allows real number co-ordinates represent theamplitude of a cosine wave, and an imaginary number represent the amplitude of thesinewave, then adding the amplitude modulated sinewave and cosine wave together forms a

'quadrature' signal.For example, COFDM is also used as the transmission scheme for DVB-T (Freeview) whichhas the option of QPSK, 16-QAM and 64-QAM signal constellations to modulate thesubcarriers. QAM stands for quadrature amplitude modulation and to generate one of thesignal points on the constellation you amplitude modulate the cosine wave and the sinewavewith the co-ordinates of the point on the signal constellation and then add the cosine waveand the sinewave together and the resultant signal is an amplitude and phase modulatedsignal, which is beneficial because you don't have to phase and amplitude modulate:

The benefit of using 16-QAM or 64-QAM is that each symbol on each subcarrier can carrymore bits of information. The number of bits that each symbol can carry is given by thefollowing equation:

number of bits = log2 M

where log2 is the logarithm to the base 2 and M is the order of the constellation. So QPSKsymbols (M=4) can carry 2 bits of information, 16-QAM symbols (M=16) can carry 4 bits ofinformation, and 64-QAM symbols (M=64) can carry 6 bits of information.

Of course, it is better to use a higher level constellation so that the overall capacity can behigher, but the drawback is that the points are closer together which makes the transmission

less robust to errors. As explained earlier, fading alters both the amplitude and phase of acarrier or subcarrier, and in the mobile channel the frequency of the subcarriers are altered by


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a Doppler shift. Also, thermal noise produced by devices in the receiver such as the RF mixeris added to the received signal, and it is this noise that is used in the signal to noise ratio(SNR) calculations.

The reason why most symbol errors occur when the signal is in a deep fade can be explainedusing the following diagram which shows how the thermal noise moves the signal point:

On DAB (using differential QPSK), if a symbol is transmitted and the subcarrier is in a deepfade then the amplitude of the subcarrier is reduced. This moves the received signal pointcloser to the origin of the diagram (co-ordinates of 0,0) and when noise is added to this in thereceiver's RF front end then because the point is already near the origin then it is easy for thenoise to move the point to a position where the difference in the angle does not fall within thedecision region allowed for a correct decision.

OFDM Modulator

After the symbol mapping is carried out, as explained above, the frequency interleaving willre-order the symbols (not shown in diagram) and then the 1536 complex numbers that

represent the symbols to be transmitted on each of the subcarriers will be sent to a serial-to-parallel converter and "placed" on each of the subcarriers. As all of this is done in the digital


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domain then the above diagram just serves as a way to visualise what happens. In reality the1536 complex numbers will be stored in two buffers, with one buffer containing the realvalues of the complex number, and the other buffer containing the imaginary values of thecomplex numbers.

The OFDM modulator consists of the block in the diagram that is labelled 'IDFT', which

stands for inverse discrete Fourier transform. Again, in reality, the actual process carried outis the inverse fast Fourier transform (IFFT), because the IFFT is, as the name suggest, a fastway to calculate the IDFT.

The IDFT calculates the following equation:

x(n) is the nth output signal complex value (time domain), X(k) is the complex symbol valueon the kth subcarrier (frequency domain), and (for DAB transmission mode TM1) N = 2048 is

the number of output signal points calculated, and also the number of input frequency points.The equation is a summation from 0 to N-1 for each output value x(n), X(k).e j.2.k.pi.n / N issummed from k=0 to k=N-1. For example, for x(2) the sum would be:

x(2) = X(0) e j.0.2.pi.2 / N + X(1) e j.1.2.pi.2 / N + X(2) e j.2.2.pi.2 / N + X(3) e j.3.2.pi.2 / N + X(4) e j.4.2.pi.2 / N +..........

To understand what the IDFT does, you first need to understand what the discrete Fouriertransform (DFT) does for which the IDFT is the inverse. The DFT calculates the discretefrequency spectrum from a block of discrete time samples of the signal (by 'discrete' I meanthat a discrete signal or discrete spectrum is only defined at discrete moments of time, e.g. atthe sampling instant for a time signal, or at a given frequency for a frequency spectrum).

Therefore, the inverse DFT calculates the discrete time samples from a discrete frequencyspectrum. This means that the frequency spectrum of the transmitted signal is given by thevalues of the complex data symbols on the subcarriers.

There are a lot of redundant operations in the DFT, and for an N-point DFT this requires N2

complex multiplications, which for example for a 2048 point DFT as would be used fortransmission mode 1 this would require 4,194,304 multiplications. The fast Fourier transform(FFT) is, as its name suggest, a fast way to calculate the DFT as many of the redundantoperations are discarded, and this allows the FFT to be calculated in (N/2) log2 Nmultiplications, which for a 2048 point FFT requires only 11,264 multiplications, which is amassive saving compared to the DFT.

One of the properties of the DFT is what makes it suitable for OFDM, and really what makesOFDM feasible for practicaly implementation in the first place. This property is that thediscrete frequency spectrum that is calculated by a DFT from a block of data samples hasfrequency samples that are all equally spaced in frequency, and this spacing equals 1/T,where T is the total duration of the time samples in the block. For example, for DABtransmission mode 1 (TM1), the "useful" duration of OFDM symbols (not data symbols onthe subcarriers, OFDM symbols carry the data symbols on the subcarriers) is 1 ms (i.e. T = 1ms), so 1/T = 1 kHz, and all the subcarriers are spaced by 1kHz. It is these equally spacedsubcarriers that equal the useful symbol duration that gives OFDM its "orthogonal" propertyin its name orthogonal frequency division multiplexing.

The property of orthogonality for communication signals means that signals that are

orthogonal to each other can be transmitted together and they don't interfere with each other.So having the subcarriers all orthogonal to one another (each subcarrier is orthogonal to all


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the other 1535 subcarriers) means that you can transmit the subcarriers in parallel and theywon't interfere with each other. This means that the individual spectra for each of thesubcarriers can overlap, and they still won't interfere with one another. A diagram that showswhat the frequency spectra of subcarriers looks like for DAB is shown below, and the numberof subcarriers for TM1 will be 1536:

:

As you can see from the figure above, for the frequency in red, all the 4 neighbouring spectraare zero where the red spectra is at its peak, and so there is no "intercarrier interference"; thisis due to the orthogonality principle.

The reason why the DFT makes OFDM practically feasible is that if you want to transmit1536 subcarriers that are all orthogonal to each other then you would need 1536 oscillatorswhich are all separated by 1kHz and 1536 filters at the transmitter, and 1536 filters andoscillators in each receiver, which is obviously not practical.

After the IFFT has been calculated, the 1536 output complex numbers are parallel to serialconverted (the P/S block in the diagram above), and following this the cyclic prefix (or guard

period) is inserted (see diagram at the start of the COFDM transmitter).

The cyclic prefix copies the complex numbers from the end of the block of output values and"pastes" them onto the front of the block (or from the front of the block copied to the end).The reason why the values from the end of the block are copied to the front is to retain

orthogonality in the multipath channel.

The reason why the end of the block is copied to the front is so that the delayed paths fromthe symbol fall within the guard period. To show why this retains orthogonality you have toconsider that the OFDM signal consists of the addition of all the subcarrier signals, which are

all at different frequencies f0 and with different values of an and bn as shown in the equationand the waveforms that are added to make the bottom OFDM signal:


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both the BBC and Digital One use the same frequency right across the UK, so the situationwhere there are multiple frequencies required is avoided.

I've found that there is a common misconception that only the BBC and Digital Onemultiplexes use the SFN concept. This is not so, and all DAB multiplexes that have morethan one transmitter for a given area use the SFN concept, and this is the vast majority of

multiplexes that I'm aware of in the UK.

COFDM Receiver

After the signals are received at the antenna, the signals are I/Q downconverted from RF togenerate the real (I) and imaginary (Q) streams, lowpass filtered (LPF) and digitized in the

analogue to digital converters (ADC, one ADC for each stream). Following the ADC, thecyclic prefix is stripped off and the remaining sampled values are serial to parallel convertedand once there is a full block of samples (1536 for TM1) the DFT is calculated (in reality theFFT is calculated as the FFT requires far fewer multiplications to be carried out than theDFT).

After the FFT (the FFT is the OFDM demodulator), the originally transmitted symbols willbe received, but they will be corrupted in that the amplitude and phase will be altered by thechannel response for each subcarrier, and noise will be added in the receiver which moves thereceived point in a random direction and with a random amplitude.

As DAB uses differential modulation, only the difference in phase between the previous andpresent symbol on each subcarrier needs to be found to decode what was sent (ignoringerrors).

The phase angle of a complex number can be found from the following formula:

theta = tan-1 (I / Q)

To find the phase difference between the previous and present symbol the complex conjugateof the previous received point is multiplied by the present received point, then the angle ofthe result of this multiplication is the phase change. The complex conjugate of a complexnumber just changes the sign of the imaginary part, for example, if you have 1 + j2, then its

complex conjugate is 1 - j2.


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Unfortunately, when DAB was specified in 1991 the engineers decided to use differentialmodulation instead of coherent (or synchronised) modulation. Synchronised modulationmeans that the absolute phase of the symbol is transmitted, rather than the difference between

phases. In 1991 differential modulation may have been seen to be a good choice, butsynchronised modulation will be used in all modern communication systems because it is

easy to synchronise the carrier, and differential modulation doubles the number of bit errorscompared to synchronised modulation. The reason for why the number of bit errors aredoubled is because if one received symbol has been rotated by the channel or by noise to anextent that it causes an error, then because the probability of a bit error is low, there is a veryhigh probability that the following symbol is also received in error.

For example, for a typical probability of error of about 0.0001, if one error occurs then theprobability that the following symbol is in error is 1-0.0001 = 0.9999, i.e. virtually certain, sooverall differential modulation doubles the number of bit errors.

Following the determination of the change of phase on each of the subcarriers, first thefrequency interleaving is reversed and then the time interleaving is reversed, and the valuesare fed into the Viterbi error correction decoder.

The output bitstream from the Viterbi decoder is then forwarded to software or hardware thatgoes about splitting the multiplexed data into its constituent streams followed by sending theaudio data to the MPEG decoder to generate the PCM bitstream that is sent to the DACs,amplified and sent to the speakers.

My Proposal to Improve DAB

Synchronous Modulation

First as I've just described, DAB uses differential modulation. It would be easy for receiversto be designed to use synchronised demodulation, and this wouldn't affect the existingreceivers that use different demodulation. This would remove the unnecessary doubling ofthe number of bit errors.

Hierarchical Modulation

As I described earlier (in the COFDM Transmitter section), the capacity of a DAB multiplexdepends on the number of points in the signal constellation. DAB uses QPSK which has 4signal points, which means that each data symbol on each subcarrier carries 2 bits of data.

Moving to a 16-QAM signal constellation would be problematic from a backwardcompatibility point of view, unless the transmitter powers were significantly increased. Butan 8-ASPK constellation would be possible:


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This scheme is called "hierarchical modulation" and a similar scheme is specified in theDVB-T (Freeview) specification. It is called hierarchical modulation because receivers with alower signal to noise ratio still receive the lower bit rate stream (called the high priority (HP)stream) while receivers with a high enough signal to noise ratio receive the higher bit ratestream (called the low priority (LP) stream).

This would increase the capacity of a DAB multiplex by 50%, and would not cause problemsin terms of backwards compatibility with existing receivers because the existing differential

phase modulation could be used, but for newer receivers with a high enough signal to noiseratio then they would be able to decode which of the two "rings" the transmitted point was

from, and hence decode the extra bit per symbol per subcarrier, which means that instead of 2symbols per subcarrier you decode 3, hence the 50% increase in capacity.

This only requires a relatively small increase in transmitter power, and the increase intransmitter power benefits the receivers that are only receiving QPSK.

In order to be backwardly compatible with existing receivers, the HP stream would have to betransmitted as it is transmitted now, while the extra capacity can be used to provide extrainformation to modify the audio bitstream in order to improve the accuracy of the audiodecoding. This would require development of additional electronics hardware, but that is atrivial task, and certainly a task worth undertaking for the reward of an extra 50% of capacityand the significantly improved audio quality that would result.

Low Density Parity Check (LDPC) Coding

LDPC codes are an old form of FEC code (invented by a famous coding theoretician calledGallagher) that have been "re-discovered" and have attracted a lot of attention from theinformation theory research community because of their near-optimum performance. Thesecodes acquire their power due to them being decoded using the so-called turbo principle,which is an iterative decoding technique from which these (and turbo codes) derive theirnear-optimum performance. FEC codes that use the turbo principle were not re-discovereduntil after DAB and DVB-T had been standardised, and so could not be used, which is ashame because FEC codes that use the turbo principle outperform the FEC coding used on

DAB by a very large margin. The two-layer FEC coding used on DVB-T (DAB uses a single


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analogue radio vs

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