COFDM

What is COFDM:
The modulation scheme that DAB uses is Coded Orthogonal Frequency Division Multiplexing
(COFDM). COFDM uses a very different method of transmission to older digital radio modulation
schemes and has been specifically designed to combat the effects of multipath 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 whereas another
path may bounce off a hill or building before reaching the aerial. In this example, the signal that
travels along the line-of-sight path arrives at the aerial first followed a short period later 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 antenna is
that the amplitude of the received signal fluctuates. The reason for this fluctuation is due to the
relative phase angle between the different paths.
The received signal is very-high frequency sinusoidal carrier signal with comparatively a very
slowly changing information signal that has been modulated onto the carrier. Therefore, a good
way to model a carrier signal 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 distances covered (the amplitude reduces the further it travels) and relative phase angle
due to the different delay.

To find out the instantaneous amplitude that is received at the antenna a vector
diagram can be drawn on which each multipath is represented by its amplitude (the length 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 (ignore the N and E)

Ignoring the labels on the above diagram, the diagram could represent a two-path signal where
the direct path is the pink vector and the sky blue vector is the delayed path, and the vector
addition produces the red vector, and it is the resultant red vector that the receiver actually
"sees".
As a mobile receiver moves relative to the transmitter the distances travelled by the paths also
changes and because the wavelength of a radio signal is of the order of 3 metres for VHF FM
signals and about 1.5 metres for DAB signals in Band III the relative phase angles between the
paths changes rapidly and randomly. For example, if there were two multipaths that are in-phase
(zero relative phase difference) then one of the paths only has to travel half a wavelength 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 blue vector had a relative phase of 180 0
and a length equal to the pink vector then it would be facing in the opposite direction to the pink
vector so the pink and sky blue vectors would completely cancel one another and the length of
the resultant red vector would be zero. As I explained above, the antenna "sees" the red vector,
so the amplitude that the antenna sees is also zero. The term for this in physics is "destructive
interference" and the signal is said to be in a "deep fade".

Deep fades occur more frequently the faster the mobile is travelling, but the duration that the
signal is in a deep fades decreases as the speed of the mobile increases. A typical graph of the
amplitude of the carrier signal that the mobile antenna sees as it travels is shown below:
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 more
than others. Narrowband signals on the other hand suffer from "flat fading" where the whole
signal 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 the
duration of the transmitted symbols over the mobile channel. The mobile channel can be
represented 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 received varies
with time due to the different multipaths that arrive at the receiver. The duration from the first
received path to the last received path that has significant power gives the maximum delay of the
channel. A typical power delay profile varies between approximately 4 µs for urban environments
up to about 20 µs for a rural environment.
A wireless digital communication system transmits "symbols" through the channel, for example,
for a single-carrier binary phase shift keying (BPSK, which uses either 0 0 phase angle or 1800
phase angles, and a carrier phase of 00 represents a bit value of 0, and a carrier phase of 180 0
represents a bit value of 1, so each transmitted "symbol" represents one bit of data) modulation
scheme then the symbol duration is the duration between when the phase angles can change.
And a wireless digital communication system uses narrowband transmission if the channel
symbol duration is greater than the maximum delay of the mobile channel (e.g. 4 µs for urban and
about 20 µs 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 when the
signal is in a deep fade. Therefore these systems must mitigate the negative effects that multipath
causes and different systems go about it in different ways. The two best known modern 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 shorter than 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 channels transmitted in parallel, so the overall spectrum
is wide (but this doesn't mean that it uses wideband transmission principles).

Error Correction Coding:

The result of OFDM using a large number of narrowband subcarriers is that each subcarrier
suffers 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 subcarriers is
protected by forward error correction (FEC) coding. The type of error correction coding that is
used in COFDM is convolutional coding and the effect of convolutional coding is that for every
one bit input to the error correction encoder, more than one bit is output depending on the "code
rate" being used. For example, a code rate of 1/3 would mean that for every bit input 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 be able to correct any bits that
are received in error. The error correction decoder used in COFDM 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 code rate
because some bits are more important for the correct reproduction of the audio than some of 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 higher
protection as more redundancy is added) so that the Viterbi error correction decoder has a higher
chance of correcting any errors.

Interleaving
Unfortunately, the Viterbi algorithm performs poorly when it is presented with bit errors that are 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 against this, DAB uses
time interleaving and frequency interleaving.
An example of how time interleaving is used is shown in the above table. The data symbols are
written into the interleaving block in column order, then once the block is full, the symbols are
read out in row order, so for example the symbols would be read out in the following 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 the symbols
to the original order.
The effect of this is to spread out symbol errors that occur grouped together. For example, as the
first few symbols transmitted in the above table would be 0, 8, 16, 24 and so on, and if a deep
fade occurs which makes symbols 8 and 16 to be received in error, then because of the re-
ordering carried out in the receiver the errors end up spread out in time, which allows the Viterbi
decoder to have a better chance of correcting all of the symbols.
As I explained in the Wideband & Narrowband Wireless Transmission section, wideband wireless
signals are subject to frequency selective fading, and because the number of subcarriers used is
large (for example DAB transmissions in the UK use Transmission Mode 1, which uses 1536
subcarriers each with a bandwidth of 1kHz), the overall DAB signal spectrum is wideband, so not
only are the narrowband subcarriers subject to flat fading, the spectrum as a whole is subject to
frequency selective fading. The result of this is that groups of neighbouring subcarriers may all be
faded. To mitigate against this, DAB uses frequency interleaving as well as time interleaving so
that after the time interleaver, the symbols read out are put on subcarriers that are a certain
distance in frequency apart. Again, the receiver reverses this and the overall effect is that the
Viterbi decoder sees the data symbols in the original order, but errors are uniformly spread out in
the stream.
Interleaving is a powerful method to improve the error correction capabilities of a wireless system
that is subject to fading, but of course it cannot perform miracles, and if too many symbols are
decoded incorrectly then it will fail and you're then likely to hear the usual "bubbling mud" sound
that is characteristic of reception problems.
COFDM Transmitter

After the bit-stream is re-ordered in the time interleaver block, 3072 bits (for Transmission Mode 1
which uses 1536 subcarriers) enter the OFDM modulator. The bit-stream is first split up into 1536
pairs of bits and each pair is mapped to one of four quaternary phase shift keying (QPSK)
symbols.
DAB uses differential QPSK, which means that the bits are mapped to phase changes rather than
to an absolute transmitted phase. An example mapping might be as follows:

Phase Change
Data Bits
(degrees)
00 0
01 90
11 180
10 270
The above mapping is called a Gray code mapping, because adjacent symbols (or in this case
phase changes) only differ by the value of one bit, which lowers the probability of there being two
bit errors for one symbol.
After the 1536 pairs of bits have been mapped to one of the four phase changes these phase
changes are applied to the 1536 subcarriers. The previously transmitted QPSK symbol on each
subcarrier will be placed in memory in the transmitter, and the phase change will then rotate this
symbol. For example, if the previous transmitted symbol on a subcarrier was the top, right-hand
point (at 45o) in the figure below (called a signal constellation diagram) and the bits that are being
mapped onto the subcarrier are '11' then the phase will rotate by 180 o so that the bottom, left-
hand point (at 225o) will be transmitted on that subcarrier.
The QPSK symbols shown in the signal constellation diagram above are represented numerically
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 are called 'complex
numbers'. A complex number consists of the combination of a real plus an imaginary number:
I+jQ

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' is that it
is equal to the square-root of -1, which doesn't actually exist, and that is why it is called an
imaginary number, but complex numbers are a very useful mathematical concept and the fact 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 processing
download 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
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 the
real and imaginary axes are at 900 apart and a sinewave and a cosine wave (both with the same
frequency) are also 900 out of phase. This allows real number co-ordinates represent the
amplitude of a cosine wave, and an imaginary number represent the amplitude of the sinewave,
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) which has
the option of QPSK, 16-QAM and 64-QAM signal constellations to modulate the subcarriers.
QAM stands for quadrature amplitude modulation and to generate one of the signal points on the
constellation you amplitude modulate the cosine wave and the sinewave with the co-ordinates of
the point on the signal constellation and then add the cosine wave and the sinewave together and
the resultant signal is an amplitude and phase modulated signal, 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 carry more
bits of information. The number of bits that each symbol can carry is given by the following
equation:
number of bits = log2 M
where log2 is the logarithm to the base 2 and M is the order of the constellation. So QPSK
symbols (M=4) can carry 2 bits of information, 16-QAM symbols (M=16) can carry 4 bits of
information, 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 be higher,
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 a carrier or
subcarrier, and in the mobile channel the frequency of the subcarriers are altered by a Doppler
shift. Also, thermal noise produced by devices in the receiver such as the RF mixer is 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 explained
using 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 deep fade
then the amplitude of the subcarrier is reduced. This moves the received signal point closer to the
origin of the diagram (co-ordinates of 0,0) and when noise is added to this in the receiver's RF
front end then because the point is already near the origin then it is easy for the noise to move
the point to a position where the difference in the angle does not fall within the decision region
allowed for a correct decision.

OFDM Modulator

After the symbol mapping is carried out, as explained above, the frequency interleaving will re-
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 domain then the above
diagram just serves as a way to visualise what happens. In reality the 1536 complex numbers will
be stored in two buffers, with one buffer containing the real values of the complex number, and
the other buffer containing the imaginary values of the complex 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 out is the inverse
fast Fourier transform (IFFT), because the IFFT is, as the name suggest, a fast way 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 value on 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.
j.2.k.pi.n / N
The equation is a summation from 0 to N-1 for each output value x(n), X(k).e is summed
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 Fourier
transform (DFT) does for which the IDFT is the inverse. The DFT calculates the discrete
frequency spectrum from a block of discrete time samples of the signal (by 'discrete' I mean that a
discrete signal or discrete spectrum is only defined at discrete moments of time, e.g. at the
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 frequency spectrum. This
means that the frequency spectrum of the transmitted signal is given by the values 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 N 2
complex multiplications, which for example for a 2048 point DFT as would be used for
transmission 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 redundant
operations are discarded, and this allows the FFT to be calculated in (N/2) log 2 N multiplications,
which for a 2048 point FFT requires only 11,264 multiplications, which is a massive saving
compared to the DFT.
One of the properties of the DFT is what makes it suitable for OFDM, and really what makes
OFDM feasible for practicaly implementation in the first place. This property is that the discrete
frequency spectrum that is calculated by a DFT from a block of data samples has frequency
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 DAB transmission mode 1 (TM1), the
"useful" duration of OFDM symbols (not data symbols on the subcarriers, OFDM symbols carry
the data symbols on the subcarriers) is 1 ms (i.e. T = 1 ms), so 1/T = 1 kHz, and all the
subcarriers are spaced by 1kHz. It is these equally spaced subcarriers that equal the useful
symbol duration that gives OFDM its "orthogonal" property in 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 the other 1535
subcarriers) means that you can transmit the subcarriers in parallel and they won't interfere with
each other. This means that the individual spectra for each of the subcarriers can overlap, and
they still won't interfere with one another. A diagram that shows what the frequency spectra of
subcarriers looks like for DAB is shown below, and the number of subcarriers for TM1 will be
1536:

As you can see from the figure above, for the frequency in red, all the 4 neighbouring spectra are
zero where the red spectra is at its peak, and so there is no "intercarrier interference"; this is due
to the orthogonality principle.
The reason why the DFT makes OFDM practically feasible is that if you want to transmit 1536
subcarriers that are all orthogonal to each other then you would need 1536 oscillators which are
all separated by 1kHz and 1536 filters at the transmitter, and 1536 filters and oscillators in each
receiver, which is obviously not practical.
After the IFFT has been calculated, the 1536 output complex numbers are parallel to serial
converted (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 from the
symbol fall within the guard period. To show why this retains orthogonality you have to consider
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 equation and the waveforms
that are added to make the bottom OFDM signal:

Then, so long as all the multipaths fall within the cyclic prefix duration and if samples are taken
over the "useful" symbol duration (as opposed to the total symbol duration that includes the cyclic
prefix) then the DFT equation that is calculated in the receiver "integrates" over an integer
number of full sinewave cycles, which is a requirement for orthogonality to hold.
Following the insertion of the cyclic prefix, the values are fed to digital to analogue converters
(DAC) and lowpass filters for each of the real and imaginary streams. The real values of the
complex numbers are then amplitude modulated onto a cosine RF (radio frequency, i.e. about
210 MHz for Band III) carrier, and the imaginary values of the complex numbers are amplitude
modulated onto a sine RF carrier. The sine and cosine carriers are then added together, and sent
through a bandpass filter and then sent to the antenna for transmission.
The insertion of the guard period between the useful symbols also enables DAB to use single-
frequency networks (SFNs):
Using a cyclic prefix means that receivers can receive signals on the same frequency from
different transmitters so long as the delay between the first and last signal to arrive falls within the
cyclic prefix duration. So signals from transmitters whose signals are delayed relative to the
signals from a closer transmitter are treated as "artificial" multipath.
SFNs allow the same frequency to be used for a given area and this means that a few low power
transmitters can be used as opposed to having one very high power transmitter. Overall, the
power required using the SFN concept is lower for transmitting to a given area. SFNs are also
spectrally efficient when it comes to frequency planning because for example, both the BBC and
Digital One use the same frequency right across the UK, so the situation where there are multiple
frequencies required is avoided.
I've found that there is a common misconception that only the BBC and Digital One multiplexes
use the SFN concept. This is not so, and all DAB multiplexes that have more than 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 to
generate 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, the cyclic
prefix is stripped off and the remaining sampled values are serial to parallel converted and once
there is a full block of samples (1536 for TM1) the DFT is calculated (in reality the FFT is
calculated as the FFT requires far fewer multiplications to be carried out than the DFT).
After the FFT (the FFT is the OFDM demodulator), the originally transmitted symbols will be
received, but they will be corrupted in that the amplitude and phase will be altered by the channel
response for each subcarrier, and noise will be added in the receiver which moves the received
point in a random direction and with a random amplitude.
As DAB uses differential modulation, only the difference in phase between the previous and
present symbol on each subcarrier needs to be found to decode what was sent (ignoring errors).
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 conjugate of
the previous received point is multiplied by the present received point, then the angle of the result
of this multiplication is the phase change. The complex conjugate of a complex number just
changes the sign of the imaginary part, for example, if you have 1 + j2, then its complex
conjugate is 1 - j2.
Unfortunately, when DAB was specified in 1991 the engineers decided to use differential
modulation instead of coherent (or synchronised) modulation. Synchronised modulation means
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, but synchronised
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 errors compared to synchronised
modulation. The reason for why the number of bit errors are doubled is because if one received
symbol has been rotated by the channel or by noise to an extent that it causes an error, then
because the probability of a bit error is low, there is a very high 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 the
probability that the following symbol is in error is 1-0.0001 = 0.9999, i.e. virtually certain, so
overall differential modulation doubles the number of bit errors.
Following the determination of the change of phase on each of the subcarriers, first the frequency
interleaving is reversed and then the time interleaving is reversed, and the values are fed into the
Viterbi error correction decoder.
The output bitstream from the Viterbi decoder is then forwarded to software or hardware that
goes about splitting the multiplexed data into its constituent streams followed by sending the
audio 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 receivers to be
designed to use synchronised demodulation, and this wouldn't affect the existing receivers that
use different demodulation. This would remove the unnecessary doubling of the number of bit
errors.

Hierarchical Modulation

As I described earlier (in the COFDM Transmitter section), the capacity of a DAB multiplex
depends on the number of points in the signal constellation. DAB uses QPSK which has 4 signal
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 backward compatibility point of view,
unless the transmitter powers were significantly increased. But an 8-ASPK constellation would be
possible:
This scheme is called "hierarchical modulation" and a similar scheme is specified in the DVB-T
(Freeview) specification. It is called hierarchical modulation because receivers with a lower 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 rate stream (called the low
priority (LP) stream).
This would increase the capacity of a DAB multiplex by 50%, and would not cause problems in
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 noise ratio 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 2 symbols per
subcarrier you decode 3, hence the 50% increase in capacity.
This only requires a relatively small increase in transmitter power, and the increase in transmitter
power benefits the receivers that are only receiving QPSK.
In order to be backwardly compatible with existing receivers, the HP stream would have to be
transmitted as it is transmitted now, while the extra capacity can be used to provide extra
information to modify the audio bitstream in order to improve the accuracy of the audio decoding.
This would require development of additional electronics hardware, but that is a trivial task, and
certainly a task worth undertaking for the reward of an extra 50% of capacity and 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 called
Gallagher) that have been "re-discovered" and have attracted a lot of attention from the
information theory research community because of their near-optimum performance. These
codes 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 their near-optimum
performance. FEC codes that use the turbo principle were not re-discovered until after DAB and
DVB-T had been standardised, and so could not be used, which is a shame 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 layer of FEC coding, and therefore the
error protection is significantly weaker than for DVB-T) is also outperformed quite significantly.
Turbo codes were the first type of FEC codes to use the turbo principle, but a major advantage of
LDPC codes are that they are far less computationally complex to decode at the receiver, and
hence the power consumption in the receiver will be less for LDPC codes than for turbo codes,
yet LDPC codes achieve approximately the same, near-optimum performance that turbo codes
achieve. It is for this reason that the DVB organisation chose LDPC codes for the new DVB-S2
digital satellite standard, which supposedly performs so close to being optimal that the DVB claim
it will never need to be replaced.
Using LDPC coding along with hierarchical modulation -- if the transmission parameters are
chosen wisely -- would allow a large percentage of people that can decode the high-priority
QPSK backwardly compatible stream to also decode the lower priority streams.

Variable Bit Rate Coding

VBR is far more efficient than constant bit rate (CBR) coding, and the Internet audio coding
community all seem to favour VBR.
Variable bit rate coding varies the bit rate on a frame-by-frame basis according to the difficulty in
encoding the audio frame so that more difficult to encode frames use a higher bit rate, while
easier to encode frames use a lower bit rate. This is a sensible way to allocate bit rate and
provides the best compression for a given audio quality. Layer 2 as used on DAB has the option
of VBR.
Using VBR allows the use of statistical multiplexing across the whole multiplex, so if the low-
priority stream was used to carry VBR information that has been statistically multiplexed then the
low-priority stream information could be used as effectively as possible to provide higher bit rates
where they’re needed and thus the audio quality across the multiplex could be dramatically
improved.
To implement this you would transmit the high-priority stream as usual with the normal bit rates,
then if the receiver can decode the low-priority stream, the VBR information could modify the
high-priority audio information to make the frequency components more accurate, and therefore
provide a higher audio quality.
This would place most of the “intelligence” at the broadcaster’s end and a receiver would just
need to have some chip or software to modify the MPEG audio data prior to the data entering the
MPEG decoder.
Overall, the scheme proposed above would allow the audio quality on DAB to be transformed
from mediocre in the extreme, to high audio quality worthy of the term "near-CD quality".
I have forwarded my proposal to someone at the BBC that can influence these things, but judging
by the present state that DAB is in then I won't hold my breath before the audio quality is any
good, and will continue to listen to Freeview until that happens.