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  Countries with regular services

  Countries with trials / tests

  Countries with interest

  DAB no longer used

Digital Audio Broadcasting (DAB) is a digital radio technology for broadcasting radio stations, used in several countries, particularly in Europe. As of 2006, approximately 1,002 stations worldwide broadcast in the DAB format.^[1]

The DAB standard was initiated as a European research project in the 1980s.^[2] The Norwegian Broadcasting Corporation (NRK) launched the very first DAB channel in the world on June 1 1995 (NRK Klassisk),^[3] and the BBC and SR launched their first DAB digital radio broadcasts in September 1995.^[4] DAB receivers have been available in many countries since the end of the nineties. DAB may offer more radio programmes over a specific spectrum than analogue FM radio. DAB is more robust with regard to noise and multipath fading for mobile listening, since DAB reception quality first degrades rapidly when the signal strength falls below a critical threshold, whereas FM reception quality degrades slowly with the decreasing signal.

An unblinded "informal listening test" by Sverre Holm has shown that for stationary listening the audio quality on DAB is subjectively lower than FM stereo (but this may be due to observer bias). Most stations using a bit rate of 128 kbit/s or less, with the MP2 audio codec, which requires 160 kbit/s to achieve perceived FM quality. 128 kbit/s gives better dynamic range or signal-to-noise ratio than FM radio, but a more smeared stereo image, and an upper cutoff frequency of 14 kHz, corresponding to 15 kHz of FM radio.^[5] However, "CD sound quality" with MP2 is possible "with 256..192 kbps".^[6]

An upgraded version of the system was released in February 2007, which is called DAB+. DAB is not forward compatible with DAB+, which means that DAB-only receivers will not be able to receive DAB+ broadcasts.^[7] DAB+ is approximately twice as efficient as DAB due to the adoption of the AAC+ audio codec, and DAB+ can provide high quality audio with as low as 64 kbit/s.^[8] Reception quality will also be more robust on DAB+ than on DAB due to the addition of Reed-Solomon error correction coding.

In spectrum management, the bands that are allocated for public DAB services, are abbreviated with T-DAB, where the "T" stands for terrestrial.

More than 20 countries provide DAB transmissions, and several countries, such as Australia, Italy, Malta, Switzerland and Germany,^[9] have started transmitting DAB+ stations. See Countries using DAB/DMB. However, DAB radio has still not replaced the old FM system in popularity.

History[link]

DAB has been under development since 1981 at the Institut für Rundfunktechnik (IRT). In 1985 the first DAB demonstrations were held at the WARC-ORB in Geneva and in 1988 the first DAB transmissions were made in Germany. Later DAB was developed as a research project for the European Union (EUREKA), which started in 1987 on initiative by a consortium formed in 1986. The MPEG-1 Audio Layer II ("MP2") codec was created as part of the EU147 project. DAB was the first standard based on orthogonal frequency division multiplexing (OFDM) modulation technique, which since then has become one of the most popular transmission schemes for modern wideband digital communication systems.

A choice of audio codec, modulation and error-correction coding schemes and first trial broadcasts were made in 1990. Public demonstrations were made in 1993 in the United Kingdom. The protocol specification was finalized in 1993 and adopted by the ITU-R standardization body in 1994, the European community in 1995 and by ETSI in 1997. Pilot broadcasts were launched in several countries in 1995.

The UK was the first country to receive a wide range of radio stations via DAB. Commercial DAB receivers began to be sold in 1999 and over 50 commercial and BBC services were available in London by 2001.

By 2006, 500 million people worldwide were in the coverage area of DAB broadcasts, although by this time sales had only taken off in the United Kingdom and Denmark. In 2006 there were approximately 1,000 DAB stations in operation world wide.^[10]

The standard was coordinated by the European DAB forum, formed in 1995 and reconstituted to the World DAB Forum in 1997, which represents more than 30 countries. In 2006 the World DAB Forum became the World DMB Forum which now presides over both the DAB and DMB standard.

In October 2005, the World DMB Forum instructed its Technical Committee to carry out the work needed to adopt the AAC+ audio codec and stronger error correction coding. This work led to the launch of the new DAB+ system.

DAB and FM/AM compared[link]

Traditionally radio programmes were broadcast on different frequencies via FM and AM, and the radio had to be tuned into each frequency, as needed. This used up a comparatively large amount of spectrum for a relatively small number of stations, limiting listening choice. DAB is a digital radio broadcasting system that through the application of multiplexing and compression combines multiple audio streams onto a relatively narrow band centred on a single broadcast frequency called a DAB ensemble.

Within an overall target bit rate for the DAB ensemble, individual stations can be allocated different bit rates. The number of channels within a DAB ensemble can be increased by lowering average bit rates, but at the expense of the quality of streams. Error correction under the DAB standard makes the signal more robust but reduces the total bit rate available for streams.

FM HD Radio versus DAB[link]

Some countries have implemented Eureka-147 Digital Audio Broadcasting (DAB). DAB broadcasts a single station that is approximately 1500 kilohertz wide (~1000 kilobits per second). That station is then subdivided into multiple digital streams of between 9 and 12 programs. In contrast FM HD Radio shares its digital broadcast with the traditional 200 kilohertz-wide channels, with capability of 300 kbit/s per station (pure digital mode). Thus HD Radio is approximately twice as data-efficient as DAB.

The first generation DAB uses the MPEG-1 Audio Layer II (MP2) audio codec which has less efficient compression than newer codecs. The typical bitrate for DAB programs is only 128 kbit/s and as a result most radio stations on DAB have a lower sound quality than FM, prompting a number of complaints.^[11] The newer FM HD Radio uses a codec based upon the MPEG-4 HE-AAC standard. Psychoacoustic studies/tests have shown HE-AAC codecs achieve the same perceived quality as MP2, but at one-quarter the bit rate (CD quality at 64 kbit/s rather than 256 kbit/s).

A directly related issue with DAB's original inefficient compression is "downgrading" stations from stereophonic to monaural, in order to include more channels into the limited 1000 kbit/s bandwidth,^[12] smaller coverage of markets as compared to analog FM, radios that are overly expensive, poor reception inside vehicles or buildings, and a general lack of interest in DAB in many countries.

Use of frequency spectrum and transmitter sites[link]

DAB gives substantially higher spectral efficiency, measured in programmes per MHz and per transmitter site, than analogue communication. This has led to an increase in the number of stations available to listeners, especially outside of the major urban areas.

Numerical example: Analog FM requires 0.2 MHz per programme. The frequency reuse factor in most countries is approximately 15, meaning that only one out of 15 transmitter sites can use the same channel frequency without problems with co-channel interference, i.e. cross-talk. Assuming a total availability of 102 FM channels at a bandwidth of 0.2MHz over the Band II spectrum of 87.5 to 108.0 MHz, an average of 102/15 = 6.8 radio channels are possible on each transmitter site (plus lower-power local transmitters causing less interference). This results in a system spectral efficiency of 1 / 15 / (0.2 MHz) = 0.30 programmes/transmitter/MHz. DAB with 192 kbit/s codec requires 1.536 MHz * 192 kbit/s / 1136 kbit/s = 0.26 MHz per audio programme. The frequency reuse factor for local programmes and multi-frequency broadcasting networks (MFN) is typically 4 or 5, resulting in 1 / 4 / (0.26 MHz) = 0.96 programmes/transmitter/MHz. This is 3.2 times as efficient as analog FM for local stations. For single frequency network (SFN) transmission, for example of national programmes, the channel re-use factor is 1, resulting in 1/1/0.25 MHz = 3.85 programmes/transmitter/MHz, which is 12.7 times as efficient as FM for national and regional networks.

Note the above capacity improvement may not always be achieved at the L-band frequencies, since these are more sensitive to obstacles than the FM band frequencies, and may cause shadow fading for hilly terrain and for indoor communication. The number of transmitter sites or the transmission power required for full coverage of a country may be rather high at these frequencies, to avoid that the system becomes noise limited rather than limited by co-channel interference. .

Sound quality[link]

The original objectives of converting to digital transmission were to enable higher fidelity, more stations and more resistance to noise, co-channel interference and multipath than in analogue FM radio. However, the leading countries in implementing DAB on stereo radio stations use compression to such a degree that it produces lower sound quality than that received from non-mobile FM broadcasts.^{[citation needed]} This is because of the bit rate levels being too low for the MPEG Layer 2 audio codec to provide high fidelity audio quality.^[13]

The BBC Research & Development department states that at least 192 kbit/s is necessary for a high fidelity stereo broadcast :

When BBC in July 2006 reduced the bit-rate of transmission of Radio 3 from 192 kbit/s to 160 kbit/s, the resulting degradation of audio quality prompted a number of complaints to the Corporation.^[15] BBC later announced that following this testing of new equipment, it would resume the previous practice of transmitting Radio 3 at 192 kbit/s whenever there were no other demands on bandwidth.

Despite the above a survey of DAB listeners (including mobile) has shown most find DAB to have equal or better sound quality than FM.^[16]

Benefits of DAB[link]

Current AM and FM terrestrial broadcast technology is well established, compatible, and cheap to manufacture. Benefits of DAB over analogue systems are explained below.

Improved features for users[link]

DAB radios automatically tune to all the available stations, offering a list for the user to select from.

DAB can carry "radiotext" (in DAB terminology, Dynamic Label Segment, or DLS) from the station giving real-time information such as song titles, music type and news or traffic updates. Advance programme guides can also be transmitted. A similar feature also exists on FM in the form of the RDS. (However, not all FM receivers allow radio stations to be stored by name.)

DAB receivers can display time of day as encoded into transmissions, so is automatically corrected when travelling between time zones and when changing to or from Daylight Saving. This is not implemented on all receivers, and some display time only when in "Standby" mode.

Some radios offer a pause facility on live broadcasts, caching the broadcast stream on local flash memory, although this function is limited.

More stations[link]

DAB is not more bandwidth efficient than analogue measured in programmes per MHz of a specific transmitter (the so called link spectral efficiency). However, it is less susceptible to co-channel interference (cross talk), which makes it possible to reduce the reuse distance, i.e. use the same radio frequency channel more densely. The system spectral efficiency (the average number of radio programmes per MHz and transmitter) is a factor three more efficient than analog FM for local radio stations, as can be seen in the above numerical example. For national and regional radio networks, the efficiency is improved by more than an order of magnitude due to the use of SFNs. In that case, adjacent transmitters use the same frequency.

In certain areas — particularly rural areas — the introduction of DAB gives radio listeners a greater choice of radio stations. For instance, in South Norway, radio listeners experienced an increase in available stations from 6 to 21 when DAB was introduced in November 2006.

Reception quality[link]

The DAB standard integrates features to reduce the negative consequences of multipath fading and signal noise, which afflict existing analogue systems.

Also, as DAB transmits digital audio, there is no hiss with a weak signal, which can happen on FM. However, radios in the fringe of a DAB signal, can experience a "bubbling mud" sound interrupting the audio and/or the audio cutting out altogether.

Due to sensitivity to doppler shift in combination with multipath propagation, DAB reception range (but not audio quality) is reduced when traveling speeds of more than 120 to 200 km/h, depending on carrier frequency.^[17]

Less pirate interference[link]

The specialised nature and cost of DAB broadcasting equipment provide barriers to pirate radio stations broadcasting on DAB. In cities such as London with large numbers of pirate radio stations broadcasting on FM, this means that some stations can be reliably received via DAB in areas where they are regularly difficult or impossible to receive on FM due to pirate radio interference.

Variable bandwidth[link]

Mono talk radio, news and weather channels and other non-music programs need significantly less bandwidth than a typical music radio station, which allows DAB to carry these programmes at lower bit rates, leaving more bandwidth to be used for other programs.

However, this had led to the situation where some stations are being broadcast in mono, see music radio stations broadcasting in mono for more details.

Transmission costs[link]

It is common belief that DAB is more expensive to transmit than FM. It is true that DAB uses higher frequencies than FM and therefore there is a need to compensate with more transmitters, higher radiated powers, or a combination, to achieve the same coverage. A DAB network is also more expensive than an FM network. However, the last couple of years has seen significant improvement in power efficiency for DAB-transmitters.

This efficiency originates from the ability a DAB network has in broadcasting more channels per network. One network can broadcast 6-10 channels (with MPEG audio codec) or 10-16 channels (with HE AAC codec). Hence, it is thought that the replacement of FM-radios and FM-transmitters with new DAB-radios and DAB-transmitters will not cost any more as opposed to newer FM facilities.^{[citation needed]}

Cheaper transmission costs is backed by independent network studies from Teracom (Sweden) and SSR/SRG (Switzerland).  Among other things they show that DAB is up to 6 times less expensive than FM.

Disadvantages of DAB[link]

Reception quality[link]

The reception quality on DAB can be poor even for people that live well within the coverage area. The reason for this is that the old version of DAB uses weak error correction coding, so that when there are a lot of errors with the received data not enough of the errors can be corrected and a "bubbling mud" sound occurs. In some cases a complete loss of signal can happen. This situation will be improved upon in the new DAB standard (DAB+, discussed below) that uses stronger error correction coding and as additional transmitters are built.

Audio Quality[link]

Broadcasters have been criticized for ‘squeezing in’ more stations per ensemble than recommended,^{[citation needed]} by:

• Minimizing the bit-rate, to the lowest level of sound-quality that listeners are willing to tolerate, such as 128 kbit/s for stereo and even 64 kbit/s for mono speech radio.
• Having few digital channels broadcasting in stereo.

Signal delay[link]

The nature of a SFN is such that the transmitters in a network must broadcast the same signal at the same time. To achieve synchronization, the broadcaster must counter any differences in propagation time incurred by the different methods and distances involved in carrying the signal from the multiplexer to the different transmitters. This is done by applying a delay to the incoming signal at the transmitter based on a timestamp generated at the multiplexer, created taking into account the maximum likely propagation time, with a generous added margin for safety. Delays in the receiver due to digital processing (e.g. deinterleaving) add to the overall delay perceived by the listener.^[18] The signal is delayed by 2–4 seconds depending on the decoding circuitry used. This has disadvantages:

• DAB radios are out of step with live events, so the experience of listening to live commentaries on events being watched is impaired;
• Delayed time signals : Even in a well-defined network with a fixed delay, the listener has to apply an offset when using the broadcast time signal to set a clock.
• Listeners using a combination of analog (AM or FM) and DAB radios (e.g. in different rooms of a house) will hear a confusing mixture when both receivers are within earshot.

Coverage[link]

As DAB is at a relatively early stage of deployment, DAB coverage is poor in nearly all countries in comparison to the high population coverage provided by FM.

Compatibility[link]

In 2006 tests began using the much improved HE-AAC codec for DAB+. Virtually none of the receivers made before 2008 support the new codec, however, thus making them partially obsolete once DAB+ broadcasts begin and completely obsolete once the old MPEG-1 Layer 2 stations are switched off. However new receivers are both DAB and DAB+ compatible.

Power requirements[link]

As DAB requires digital signal processing techniques to convert from the received digitally encoded signal to the analogue audio content, the complexity of the electronic circuitry required to do this is high. This translates into needing more power to effect this conversion than compared to an analogue FM to audio conversion, meaning that portable receiving equipment will tend to have a shorter battery life, or require higher power (and hence more bulk). This means that they use more energy than analogue Band II VHF receivers.

As an indicator of this increased power consumption, some radio manaufacturers quote the length of time their receivers can play on a single charge. For a commonly used FM/DAB-receiver from manufacturer PURE, this is stated as: DAB 10 hours, FM 22 hours.^{[citation needed]}

Use of Licensed Codecs[link]

The use of MPEG and latterly AAC has prompted criticism of the fact that a (large) public system is financially supporting a private company. In general, an open system will permit equipment to be bought from various sources in competition with each other but by selecting a single vendor of codec, with which all equipment must be compatible, this is not possible.

Technology[link]

Bands and modes[link]

DAB uses a wide-bandwidth broadcast technology and typically spectra have been allocated for it in Band III (174–240 MHz) and L band (1452–1492 MHz), although the scheme allows for operation almost anywhere above 30 MHz. The US military has reserved L-Band in the USA only, blocking its use for other purposes in America, and the United States has reached an agreement with Canada that the latter will restrict L-Band DAB to terrestrial broadcast to avoid interference.^{[citation needed]}

DAB has a number of country specific transmission modes (I, II, III and IV). For worldwide operation a receiver must support all 4 modes:

• Mode I for Band III, Earth
• Mode II for L-Band, Earth and satellite
• Mode III for frequencies below 3 GHz, Earth and satellite
• Mode IV for L-Band, Earth and satellite

Protocol stack[link]

From a OSI model protocol stack viewpoint, the technologies used on DAB inhabit the following layers: the audio codec inhabits the presentation layer. Below that is the data link layer, in charge of packet mode statistical multiplexing and frame synchronization. Finally, the physical layer contains the error-correction coding, OFDM modulation, and dealing with the over-the-air transmission and reception of data. Some aspects of these are described below.

Audio codec[link]

The older version of DAB that is being used in Denmark, Ireland*, Norway, Switzerland* and the UK, uses the MPEG-1 Audio Layer 2 audio codec, which is also known as MP2 due to computer files using those characters for their file extension. (*Both Ireland and Switzerland also use DAB+).

The new DAB+ standard has adopted the HE-AAC version 2 audio codec, commonly known as AAC+ or aacPlus. AAC+ is approximately three-times more efficient than MP2,^[19] which means that broadcasters using DAB+ will be able to provide far higher audio quality or far more stations than they can on DAB, or, as is most likely, a combination of both higher audio quality and more stations will be provided.

One of the most important decisions regarding the design of a digital radio system is the choice of which audio codec to use, because the efficiency of the audio codec determines how many radio stations can be carried on a multiplex at a given level of audio quality. The capacity of a DAB multiplex is fixed, so the more efficient the audio codec is, the more stations can be carried, and vice versa. Similarly, for a fixed bit-rate level, the more efficient the audio codec is the higher the audio quality will be.

Error-correction coding[link]

Error-correction coding (ECC) is an important technology for a digital communication system because it determines how robust the reception will be for a given signal strength - stronger ECC will provide more robust reception than a weaker form.

The old version of DAB uses punctured convolutional coding for its ECC. The coding scheme uses unequal error protection (UEP), which means that parts of the audio bit-stream that are more susceptible to errors causing audible disturbances are provided with more protection (i.e. a lower code rate) and vice versa. However, the UEP scheme used on DAB results in there being a grey area in between the user experiencing good reception quality and no reception at all, as opposed to the situation with most other wireless digital communication systems that have a sharp "digital cliff", where the signal rapidly becomes unusable if the signal strength drops below a certain threshold. When DAB listeners receive a signal in this intermediate strength area they experience a "burbling" sound which interrupts the playback of the audio.

The new DAB+ standard has incorporated Reed-Solomon ECC as an "inner layer" of coding that is placed around the byte interleaved audio frame but inside the "outer layer" of convolutional coding used by the older DAB system, although on DAB+ the convolutional coding uses equal error protection (EEP) rather than UEP since each bit is equally important in DAB+. This combination of Reed-Solomon coding as the inner layer of coding, followed by an outer layer of convolutional coding - so-called "concatenated coding" - became a popular ECC scheme in the 1990s, and NASA adopted it for its deep-space missions. One slight difference between the concatenated coding used by the DAB+ system and that used on most other systems is that it uses a rectangular byte interleaver rather than Forney interleaving in order to provide a greater interleaver depth, which increases the distance over which error bursts will be spread out in the bit-stream, which in turn will allow the Reed-Solomon error decoder to correct a higher proportion of errors.

The ECC used on DAB+ is far stronger than is used on DAB, which, with all else being equal (i.e. if the transmission powers remained the same), would translate into people who currently experience reception difficulties on DAB receiving a much more robust signal with DAB+ transmissions. It also has a far steeper "digital cliff", and listening tests have shown that people prefer this when the signal strength is low compared to the shallower digital cliff on DAB.^[19]

Modulation[link]

Immunity to fading and inter-symbol interference (caused by multipath propagation) is achieved without equalization by means of the OFDM and DQPSK modulation techniques. For details, see the OFDM system comparison table.

Using values for the most commonly used transmission mode on DAB, Transmission Mode I (TM I), the OFDM modulation consists of 1,536 subcarriers that are transmitted in parallel. The useful part of the OFDM symbol period is 1 millisecond, which results in the OFDM subcarriers each having a bandwidth of 1 kHz due to the inverse relationship between these two parameters, and the overall OFDM channel bandwidth is 1,537 kHz. The OFDM guard interval for TM I is 246 microseconds, which means that the overall OFDM symbol duration is 1.246 milliseconds. The guard interval duration also determines the maximum separation between transmitters that are part of the same single-frequency network (SFN), which is approximately 74 km for TM I.

Single-frequency networks[link]

OFDM allows the use of single-frequency networks (SFN), which means that a network of transmitters can provide coverage to a large area - up to the size of a country - where all transmitters use the same transmission frequency. Transmitters that are part of an SFN need to be very accurately synchronised with other transmitters in the network, which requires the transmitters to use very accurate clocks.

When a receiver receives a signal that has been transmitted from the different transmitters that are part of an SFN, the signals from the different transmitters will typically have different delays, but to OFDM they will appear to simply be different multipaths of the same signal. Reception difficulties can arise, however, when the relative delay of multipaths exceeds the OFDM guard interval duration, and there are frequent reports of reception difficulties due to this issue when there is a lift, such as when there's high pressure, due to signals travelling farther than usual, and thus the signals are likely to arrive with a relative delay that is greater than the OFDM guard interval.

Low power gap-filler transmitters can be added to an SFN as and when desired in order to improve reception quality, although the way SFNs have been implemented in the UK up to now they have tended to consist of higher power transmitters being installed at main transmitter sites in order to keep costs down.

Bit rates[link]

An ensemble has a maximum bit rate that can be carried, but this depends on which error protection level is used. However, all DAB multiplexes can carry a total of 864 "capacity units". The number of capacity units, or CU, that a certain bit-rate level requires depends on the amount of error correction added to the transmission, as described above. In the UK, most services transmit using 'protection level three', which provides an average ECC code rate of approximately ½, equating to a maximum bit rate per multiplex of 1184 kbit/s.

Services and ensembles[link]

Various different services are embedded into one ensemble (which is also typically called a multiplex). These services can include:

• Primary services, like main radio stations
• Secondary services, like additional sports commentaries
• Data services
  • Electronic Programme Guide (EPG)
  • Collections of HTML pages and digital images (Known as 'Broadcast Web Sites')
  • Slideshows, which may be synchronised with audio broadcasts. For example, a police appeal could be broadcast with the e-fit of a suspect or CCTV footage.
  • Video
  • Java Platform Applications
  • IP tunneling
  • Other raw data

[edit] DAB⁺ and DMB

The term DAB most commonly refers both to a specific DAB-standard using the MP2 audio codec, but can sometimes refer to a whole family of DAB related standards, such as DAB+, DMB and DAB-IP.

[edit] DAB⁺

WorldDMB, the organisation in charge of the DAB standards, announced DAB+, a major upgrade to the DAB standard in 2006, when the HE-AAC v2 audio codec^[20] (also known as eAAC+) was adopted. The new standard, which is called DAB+, has also adopted the MPEG Surround audio format and stronger error correction coding in the form of Reed-Solomon coding. DAB+ has been standardised as ETSI TS 102 563.

As DAB is not forward compatible with DAB+, older DAB receivers can not receive DAB+ broadcasts. However, DAB receivers that will be able to receive the new DAB+ standard via a firmware upgrade went on sale in July 2007. If a receiver is DAB+ compatible, there will be a sign on the product packaging.

DAB+ broadcasts have launched in several countries like Switzerland,^[21] Malta, Ireland, Italy, Australia and Germany. Malta was the first country to launch DAB+ in Europe. Several other countries are also expected to launch DAB+ broadcasts over the next few years, such as Hungary and Asian countries, such as China and Vietnam. such as Japan and the U.S. If DAB+ stations launch in established DAB countries, they can transmit alongside existing DAB stations that use the older MPEG-1 Audio Layer II audio format, and most existing DAB stations are expected to continue broadcasting until the vast majority of receivers support DAB+.^[22]

DMB[link]

Digital Multimedia Broadcasting (DMB) and DAB-IP are suitable for mobile radio and TV both because they support MPEG 4 AVC and WMV9 respectively as video codecs. However, a DMB video subchannel can easily be added to any DAB transmission—as DMB was designed from the outset to be carried on a DAB subchannel. DMB broadcasts in Korea carry conventional MPEG 1 Layer II DAB audio services alongside their DMB video services.

Norway, South Korea and France are countries currently broadcasting DMB. So the digital communication is very important for quality sound.

Countries using DAB[link]

More than 30 countries provide DAB, DAB+ and/or DMB broadcasts, either as a permanent technology or as test transmissions.

See also[link]

• ATSC Standards
• Digital Audio Radio Service
• Digital Multimedia Broadcasting
• Digital radio
• Digital television
• Digital Radio Mondiale
• ETSI Satellite Digital Radio (SDR)
• FMeXtra
• HD Radio
• Internet radio device
• ISDB
• OpenCable
• Orthogonal frequency-division multiplexing
• Satellite Radio
• Sirius Satellite Radio
• Spectral efficiency comparison table
• T-DMB
• Teletext
• XM Satellite Radio

References[link]

• ETSI Specifications available at ETSI Publications Download Area, pda.etsi.org (this will open ETSI document search engine, to find the latest version of the document enter a search string; free registration is required to download PDF)
• Stott, J. H.; The How and Why of COFDM, BBC Research Development

External links[link]

This article's use of external links may not follow Wikipedia's policies or guidelines. Please improve this article by removing excessive or inappropriate external links, and converting useful links where appropriate into footnote references. (March 2012)

• ETSI EN 300 401 v1.4.1 - Original DAB specification, etsi.org
• ETSI TS 102 563 v1.1.1 - DAB+ enhancement specification, etsi.org
• World DAB Forum, wordlddab.org
• DAB Ensembles Worldwide (also known as "Wohnort", the main part of the site is a list of services currently transmitting), wohnort.org
• Ofcom – UK broadcasting regulators, ofcom.org.uk
• Digital One - The UK's Commercial Digital Radio Network, ukdigitalradio.com
• DAB Digital Radio News and Information for the UK, uk-dab.info
• UK DAB sceptical news and information, digitalradiotech.co.uk
• Audio extracts of UK DAB stations, dab.blip.tv
• Digital broadcasting in Ireland, rte.i.e.
• DAB in the Netherlands, dabtuners.nl
• E-magazine with information about DAB, digitaleradio.nl
• BBC Datasheet on Eureka 147, bbd.co.uk
• ASA Bans DAB Audio Quality Claims, digitalradiotech.co.uk
• DAB and Mobile Java Services, ebu.ch
• MMBTools, Linux based free/open DAB, DAB+, T-DMB transmission infrastructure (software based), crc.ca
• OpenMokast, free/open software receiver for DAB, DAB+ and T-DMB, openmokast.org
• Opendigitalradio.org, experimentation and techniques for DAB transmission and reception, opendigitalradio.org
• DAB Car Radio, edabcarradio.co.uk

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A sound wave, in gray, represented digitally, in red (after quantization and zero-order hold).

Digital audio is sound reproduction using pulse-code modulation and digital signals. Digital audio systems include analog-to-digital conversion (ADC), digital-to-analog conversion (DAC), digital storage, processing and transmission components. A primary benefit of digital audio is in its convenience of storage, transmission and retrieval.

Overview of digital audio[link]

Sampling and 4-bit quantization of an analog signal (red) using Pulse-code modulation

Digital audio is useful in the recording, manipulation, mass-production, and distribution of sound. Modern distribution of music across the Internet via on-line stores depends on digital recording and digital compression algorithms. Distribution of audio as data files rather than as physical objects has significantly reduced the cost of distribution.

In an analog audio system, sounds begin as physical waveforms in the air, are transformed into an electrical representation of the waveform, via a transducer (for example, a microphone), and are stored or transmitted. To be re-created into sound, the process is reversed, through amplification and then conversion back into physical waveforms via a loudspeaker. Although its nature may change, analog audio's fundamental wave-like characteristics remain the same during its storage, transformation, duplication, and amplification.

Analog audio signals are susceptible to noise and distortion, unavoidable due to the innate characteristics of electronic circuits and associated devices. In the case of purely analog recording and reproduction, numerous opportunities for the introduction of noise and distortion exist throughout the entire process. When audio is digitized, distortion and noise are introduced only by the stages that precede conversion to digital format, and by the stages that follow conversion back to analog.

The digital audio chain begins when an analog audio signal is first sampled, and then (for pulse-code modulation, the usual form of digital audio) it is converted into binary signals—‘on/off’ pulses—which are stored as binary electronic, magnetic, or optical signals, rather than as continuous time, continuous level electronic or electromechanical signals. This signal may then be further encoded to allow correction of any errors that might occur in the storage or transmission of the signal, however this encoding is for error correction, and is not strictly part of the digital audio process. This "channel coding" is essential to the ability of broadcast or recorded digital system to avoid loss of bit accuracy. The discrete time and level of the binary signal allow a decoder to recreate the analog signal upon replay. An example of a channel code is Eight to Fourteen Bit Modulation as used in the audio Compact Disc (CD).

Conversion process[link]

The lifecycle of sound from its source, through an ADC, digital processing, a DAC, and finally as sound again.

A digital audio system starts with an ADC that converts an analog signal to a digital signal.^{[note 1]} The ADC runs at a sampling rate and converts at a known bit resolution. For example, CD audio has a sampling rate of 44.1 kHz (44,100 samples per second) and 16-bit resolution for each channel. For stereo there are two channels: 'left' and 'right'. If the analog signal is not already bandlimited then an anti-aliasing filter is necessary before conversion, to prevent aliasing in the digital signal. (Aliasing occurs when frequencies above the Nyquist frequency have not been band limited, and instead appear as audible artifacts in the lower frequencies).

The digital audio signal may be stored or transmitted. Digital audio storage can be on a CD, a digital audio player, a hard drive, USB flash drive, CompactFlash, or any other digital data storage device. The digital signal may then be altered in a process which is called digital signal processing where it may be filtered or have effects applied. Audio data compression techniques — such as MP3, Advanced Audio Coding, Ogg Vorbis, or FLAC — are commonly employed to reduce the file size. Digital audio can be streamed to other devices.

The last step is for digital audio to be converted back to an analog signal with a DAC. Like ADCs, DACs run at a specific sampling rate and bit resolution but through the processes of oversampling, upsampling, and downsampling, this sampling rate may not be the same as the initial sampling rate.

History of digital audio use in commercial recording[link]

Pulse-code modulation was invented by British scientist Alec Reeves in 1937^[1] and was used in telecommunications applications long before its first use in commercial broadcast and recording. Commercial digital recording was pioneered in Japan by NHK, and Nippon Columbia (a.k.a. Denon) in the 1960s. The first commercial digital recordings were released in 1971.^[2]

The BBC also began experimenting with digital audio in the 1960s. By the early 1970s they had developed a 2-channel recorder and in 1972 they deployed a digital audio transmission system linking their broadcast center to their remote transmitters.^[2]

The first 16-bit PCM recording in the United States was made by Thomas Stockham at the Santa Fe Opera in 1976 on a Soundstream recorder. In 1978, an improved version of the Soundstream system was used by Telarc to produce several classical recordings. At the same time 3M was well along in development of their digital multitrack recorder based on BBC technology. The first all-digital album recorded on this machine was Ry Cooder's "Bop 'Til You Drop" which was released in 1979. In a crash program started in 1978, British record label Decca developed their own 2-track digital audio recorders. Decca released the first European digital recording in 1979.^[2]

Helped along by introduction of popular digital multitrack recorders from Sony and Mitsubishi in the early 1980s, digital recording was soon embraced by the major record companies. With the introduction of the CD by Sony and Philips in 1982, digital audio was embraced by consumers as well.^[2]

Digital audio technologies[link]

Digital audio broadcasting

• Digital Audio Broadcasting (DAB)
• HD Radio
• Digital Radio Mondiale (DRM)
• In-band on-channel (IBOC)

Storage technologies

• Digital audio player
• Digital Audio Tape (DAT)
• Compact Disc (CD)
• Hard disk recorder
• DVD Audio
• MiniDisc
• Super Audio CD
• Various audio file formats

Digital audio interfaces[link]

Audio-specific interfaces include:

• AC'97 (Audio Codec 1997) interface between Integrated circuits on PC motherboards
• Intel High Definition Audio A modern replacement for AC'97
• ADAT interface
• AES3 interface with XLR connectors
• AES47, Professional AES3-style digital audio over Asynchronous Transfer Mode networks
• I²S (Inter-IC sound) interface between Integrated circuits in consumer electronics
• MADI Multichannel Audio Digital Interface
• MIDI low-bandwidth interconnect for carrying instrument data; cannot carry sound but can carry digital sample data in non-realtime
• S/PDIF, either over coaxial cable or TOSLINK
• TDIF, TASCAM proprietary format with D-sub cable
• A2DP via Bluetooth

Naturally, any digital bus (e.g., USB, FireWire, and PCI) can carry digital audio. Also, several interfaces are engineered to carry digital video and audio together, including HDMI and DisplayPort.

In professional architectural or installation applications, many digital audio networking protocols and interfaces exist.

See also[link]

• Analog recording vs. digital recording
• Digital audio editor
• Digital audio workstation
• Digital signal processing
• Noise shaping
• Nyquist–Shannon sampling theorem
• Quantization (signal processing)
• HTML5 Audio
• Sampling (signal processing)

Notes[link]

1. ^ Some audio signals such as those created by digital synthesis originate entirely in the digital domain, in which case analog to digital conversion does not take place.

References[link]

Sources[link]

• Borwick, John, ed., 1994: Sound Recording Practice (Oxford: Oxford University Press)
• Bosi, Marina, and Goldberg, Richard E., 2003: Introduction to Digital Audio Coding and Standards (Springer)
• Ifeachor, Emmanuel C., and Jervis, Barrie W., 2002: Digital Signal Processing: A Practical Approach (Harlow, England: Pearson Education Limited)
• Rabiner, Lawrence R., and Gold, Bernard, 1975: Theory and Application of Digital Signal Processing (Englewood Cliffs, New Jersey: Prentice-Hall, Inc.)
• Watkinson, John, 1994: The Art of Digital Audio (Oxford: Focal Press)

Ida Maria
Ida Maria playing guitar in 2007
Background information
Also known as	Ida Maria Børli Sivertsen
Born	(1984-07-13) 13 July 1984 (age 27)
Origin	Nesna, Norway
Genres	Punk rock Indie rock Alternative rock
Instruments	Guitar, Bass, Vocals
Years active	2007-
Labels	Nesna Records/Mercury/Lava
Website	www.idamaria.co.uk

Ida Maria Børli Sivertsen (born 13 July 1984), is a Norwegian rock musician and songwriter known primarily by the stage name Ida Maria, (pronounced 'Ee-da'). Her first album, Fortress Round My Heart was released in 2008. In November 2010 she released her second album Katla in Norway; the album was released internationally on 7 June 2011.

Biography[link]

Ida Maria was born and raised in the small town of Nesna in Helgeland, Norway]. She currently lives in Nesna.^{[citation needed]}

She specializes in fiercely energetic and exciting alternative pop-punk music with catchy melodies and witty lyrics. Ida taught herself to play guitar and began performing at local concerts at age fourteen.^[1]

Ida Maria gained considerable success in Norway in 2007 after winning two national competitions for unknown artists - Zoom urørt 2006 and Urørtkonkurransen 2007 - and playing successful shows at the annual Norwegian music industry festival By:Larm 2007 and 2008. Her singles "Oh My God" and "Stella" were both played regularly on Norwegian national radio station NRK P3.^{[citation needed]}

Ida Maria later expanded her popularity outside of her native country, namely in the United Kingdom, where she appeared on Later... with Jools Holland, was interviewed by The Times, and performed at the Glastonbury Festival.^[2]

In September, 2009 Ida released a remix of her debut single "Oh My God" with punk rock icon Iggy Pop singing backing vocals.^[3]

Having performed at various festivals, including Reading, and being nominated for Best Rock/Indie Artist category at the BT Digital Music Awards 2008, Ida Maria has toured in Scandinavia, the United Kingdom, the United States and Australia.^[4]

Ida Maria performed her song "I Like You So Much Better When You're Naked" on The Tonight Show with Jay Leno in 2009.^[5]

According to her Twitter account, she is currently recording her third album.

Synaesthesia[link]

Ida Maria is reported to have the neurological phenomenon of synesthesia which, in her case, means she envisions colours when she hears music.^[6]

‘I don’t know what life’s like without it, as I've always had it,’ she says. ‘As I’ve gotten older, I’ve seen that it has definitely been a bonus in terms of what I do. When I started to play guitar, I made this colour sheet for all the chords. I didn’t know what they were called, but I knew exactly what colours and shades they were. So I made songs with colour palettes which was really cool.’^[7]

Soundtrack[link]

Ida Maria's song "I Like You So Much Better When You're Naked" was featured in episode 8 of My Life As Liz entitled "A Prom to Remember Part 2". It was played during the dancing sequence at the prom. Her song "Louie" is being used for some commercials for the FX TV show Louie. The song "Oh My God" was one of the first songs confirmed to be featured in Rock Band 3.^[8]

Ida Maria's song "Keep Me Warm" was featured in the season 4 finale of the ABC drama Grey's Anatomy and during some Nivea commercials during the Fall of 2009.^[9]

Ida Maria's song "Oh My God" was featured in the trailer for It's Kind of a Funny Story and The Virginity Hit.^[10]

Ida Maria was one of the co-headliners of the 2009 Perez Hilton Presents tour. On 17 September 2009, after a Boston show, she pulled out of the tour, citing exhaustion.^[11]

Her second album, Katla, was released on 8 November 2010 in Norway.^[12] And internationally on 7 June 2011.

Her song "Bad Karma" was featured in the 2011 film Scream 4 and episodes of Gossip Girl, Teen Wolf and Being Human together with the song "Quite Nice People", both from the album Katla.

Ida Maria's song "Oh My God" was featured in a Finnish film, Sisko tahtoisin jäädä (Sister, I wanted to stay), released in 2010.

Collaborations[link]

In 2008 Ida Maria contributed backing vocals for the track "Words" on Guillemots' 2008 album Red.^{[citation needed]}

In 2009 Ida Maria collaborated with the Norwegian band DeLillos for the track "Flink" on DeLillos' 2009 album Huskeglemme ^[13]

Ida Maria has collaborated with the Norwegian rapper Joddski on two occasions, first on the track "Sannhetsserum" on Joddski's 2010 album Krig & Kjærlighet,^[14] second time on the track "Hold Ut" on Joddski's 2011 album Bensin På Bålet^[15]

In 2012 Ida Maria did a duet with Anthony Green for the track "Can't Be Satisfied" on Anthony Green's 2012 album Beautiful Things^[16]

Equipment[link]

On stage Ida Maria prefers a white Fender Stratocaster American Vintage 62 reissue together with a Fender DeLuxe amp.^[17]

Discography[link]

Studio albums

• 2008 – Fortress Round My Heart produced by Timo Räisänen - UK Number 39^[18]
• 2010 - Katla produced by Butch Walker

Singles

• "Oh My God"
• "Drive Away My Heart"
• "Stella" - UK Number 185^[19]
• "Queen of the World"
• "I Like You So Much Better When You're Naked" - UK Number 13 & 19 in Norway
• "Oh My God" (UK re-release 26 January.) UK Number 85
• "Bad Karma" (Norwegian release 11 September 2010 - International release 3 May 2011) ^[20]
• "Cherry Red" (2011)

Band[link]

For her first album, Fortress Round My Heart, Ida Maria's backing band included:^[21]

• Ida Maria Børli Sivertsen – vocals, rhythm guitar
• Stefan Törnby – lead guitar
• Johannes Lindberg – bass guitar
• Dave Krusen – drums

References[link]

• http://www.musicweek.com/story.asp?sectioncode=1&storycode=1036748

External links[link]

• Official website
• Times Online - Ida Maria interview