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18th May 2008

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How DAB works

DAB digital radio uses a transmission system known as coded orthogonal frequency division multiplex (COFDM). This divides the signals amongst several hundred low bit rate carriers.

Because the bit rates are low, the signal can be subject to large multipath dispersion without any adverse effect on reception. In fact, multipath reinforces and improves reception.

For DAB, the bit rates on the individual carriers are low enough for different transmitters carrying the same programmes to broadcast on the same frequency, provided they are correctly synchronised. This enables much more efficient use of the radio spectrum.

 

Each DAB transmission, known as a multiplex, is spread over about 1.5 MHz, with 4 phase (2 bit) symbols. This gives a data rate of about 2.4 Mbit/s, about half of which is used for error correction (though this may be varied). Each multiplex carries between 6 and 11 radio stations, depending on the bit rate (and hence sound quality) allocated to each station. The data for all the stations is distributed amongst all of the carriers. This is to ensure that interference to a few carriers does not cause more disruption to a particular station's bit stream than the error correction software can handle. Time interleaving is also used to protect against short duration interference spikes.

A more detailed description of COFDM transmission can be found in a BBC conference paper (hosted by the BBC).

A CD codes audio at a rate of about 1.2 Mbit/s. Using the same coding standard on DAB would only allow one station per multiplex - not very practical. Therefore audio compression is used to reduce the data rate of each station. Audio compression technqiues all make use of psycho-acoustic coding. This takes advantage of the fact that when a sound on one frequency is heard, the ear is rendered much less sensitive to quieter sounds on nearby frequencies. A psycho-acoustic coder performs a Fourier analysis of the sound to be coded and calculates a noise floor. Sound below the noise floor can not be perceived by the listener, so can be discarded. Only the sound above the noise floor need be calculated and this requires much fewer bits than coding the whole sound, as is done on CDs.

The audio coding standard used for DAB is MPEG (Motion Picture Expert Group) 2 Layer 2, abbreviated to MP2. This samples at 48k samples per second (twice the maximum audio frequency). Each block of 384 samples is broken into 32 equal frequency bands of 12 samples each and a separate noise floor and scale factor is set for each band. Noise floor and scale factor information is shared between sets of three blocks to save bits. Stereo stations mostly use a technique known as joint stereo whereby lower bit rate stereo separation data is added to a mono signal rather than coding the left and right channels in full.

A more detailed description of audio coding by David Robinson can be found here (569 kB, hosted on Digital Radio Tech).


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