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Hi, welcome to module 9.3 of digital
signal processing.

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We're still talking about digital
communication systems.

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In the previous module we addressed the
bandwidth constraint,

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and in this module we will tackle the
power constraint.

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So first we will introduce the concept of

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noise and probability of error in a
communication system.

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We will look at signalling alphabet and
the related power.

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And finally we'll introduce QAM
signalling.

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So, we have seen the transmitter

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sends a sequence of symbols, a of n,
created by the mapper.

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Now we take the receiver into account.

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We don't yet know how, but it's safe to
assume

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that the receiver in the end, will obtain
an estimation.

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Hat a of n, of the original transmitted
symbol sequence.

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It's an estimation.because even if there
is no distortion

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introduced by the channel, even if nothing
bad happens.

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There will always be a certain amount

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of noise that will corrupt the original
sequence.

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When noise is very large, our estimate for
the transmitted

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symbol will be off and will incur a
decoding error.

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Now this probability of error will depend
on the power of

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the noise with respect to the power of the
signal and

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will also depend on the decoding
strategies that we put in

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place, how smart we are in circumvent and
defects of the noise.

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One way we can maximize the probabilty of

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correctly guessing the transmitted symbol
is by using

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suitable alphabets.

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And so we'll see in more detail what that
means.

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Remember the scheme for the transmitter.

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We have a bitstream coming in and then we
have the scrambler,

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and then the mapper.
And here we have a sequence of symbols

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a of n.
These symbols will have to be sent over

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the channel and to do so we up sample and
we interpolate and

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then we transmit.

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Now, how do we go from bitstreams to
samples in more detail.

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In other words, how does the mapper work?

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The mapper will split the incoming
bitstreams into chunks.

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And will assign a symbol, a of n, from a
finite alphabet to each chunk.

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The alphabet we will decide later what it
is composed of.

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To undo the mapping operation and recover
the bitstream.

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The receiver will perform a slicing
operation.

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So, the receiver will receive a value of
hat a of n where

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hat indicates the fact that noise has
leaked into the value of the signal.

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And the receiver will decide which symbol
from the alphabet

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which is known to the receiver as well is
closest

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to the received symbol, and from there, it
will be

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extremely easy to piece back the original
bitstream as an example,

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let's look at simple two level signalling.

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This generates signals of the kind we has
seen

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in the examples so far alternating between
two levels.

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The way the mapper works is by splitting
the incoming bitstream into single bits.

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And the output symbol sequence uses an
alphabet composed of two symbols

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G and minus G, and associates G to bit
value

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one, and minus G to a bit of value zero.

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At the receiver the slicer, looks at the
sign of

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the incoming symbol sequence which has
been corrupted by noise

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and decides that the nth bit will be one,
if

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the sign of the nth symbol is positive and
zero otherwise.

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Let's look at an example, let's assume g
equal to 1,

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so the two level signal will alternate
between plus 1 and

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minus 1.

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And suppose we have an input bit sequence
that gives rise to this signal here.

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After transmission and after decoding at
the receiver.

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The resulting symbol sequence will look
like this, where each

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symbol has been corrupted by a varying
amount of noise.

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If we now slice this sequence by
thresholding, as

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shown before, we recover a symbol sequence
like this.

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Where we

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have indicated in red the errors incurred
by the slicer because of the noise.

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So if you want to analyze in more detail
what the probability of error is.

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We have to make some hypothesis on the
signals involved in this toy experiment.

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Assume that each received symbol can be
modelled

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as the original symbol plus a noise
sample.

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Assume also that the bits in the bits
stream are equiprobable.

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So zero and one

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appear with probability 50% each.

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Assume that the noise and the signal are
independent and assume that the

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noise is additive white Gaussian noise
with

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zero mean and known variance, sigma zero.

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With this hypothesis the probability of
error can be written out as follows.

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First of all, we split the probability

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of errors into two conditional
probabilities conditioned by

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whether the n-th bit is equal to one or
the n-th bit is equal to zero.

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In the first case when the n-th bit is
equal

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to one, remember the produce symbol will
be equal to G.

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So the probability of error is equal to
the probability

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for the noise sample to be less than minus
G.

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Because only in this case, the sum of the
sample plus the noise, will be negative.

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Similarly, when the amplitude is equal to
zero, we have a negative sample.

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And the only way for that to change sine
is if the noise

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sample is greater than G.

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Since the probability of each occurrence
is one

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half, because of the symmetry of the
Gaussian

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distribution function, this is equal to
the probability

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for the noise sample to be larger than G.

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And we can compute this as the integral
from G to infinity of

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the probability distribution function for
the

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Gaussian distribution with the known
variance here.

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So what we have here is the

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tail probability of a Gaussian with
standard deviation sigma zero.

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So, this is the integral that we're trying
to compute.

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Usually the tail probability of a unit
variance

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Gaussian is indicated by the notation Q of
G.

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The Q function of G.

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Now because our variance is actually sigma
zero squared, we

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have to normalize the Q function argument
by the standard deviation.

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And we find that the probability

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of error is equal to the Q function of G
over sigma 0.

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Since this integral is not computable
exactly, if we want to calculate the Q

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function, we have to resort to numerical

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packages, or tabulated versions of this
function.

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And usually, what you would find in

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numerical packages in, is a derived
function

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called the error function, which is
related

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to the Q function by this formula here.

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But the point that is really important

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at the end of this derivation is that the
probability of error is equal to some

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function of the ratio between the
amplitude of

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the signal, and the standard deviation of
the noise.

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Now we can carry this analysis further by
considering the transmitted power.

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We have a bi-level signal, and each level
occurs with one half probability.

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So the variance of the signal, which
corresponds

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to the power is equal to G squared times

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the probability of the n-th bit being
equal to 1, plus G squared times the

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probability of the n-th bit being equal to
0, which is equal to G squared.

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And so if we rewrite the probability of
error, we can see that it is equal to the

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Q function of the ratio between the
standard deviation

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of the signal and the standard deviation
of the noise.

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But this is really equivalent to saying
that the probability of error is equal

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to the Q function of the square root of
the signal to noise ratio of

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the transmitted signal.

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If we plot this as a function of the
signal noise to ratio in dBs and I remind

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here that dBs here mean that we compute 10
times

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the log in base 10 of the power of the
signal divided by the power of the noise.

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And since we are in a log, log scale.

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We can see that the probability of error
decays exponentially with the signal to

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noise ratio.

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This exponentially decay is quite a norm
in communication systems

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and while the absolutely rate of decay
might change in terms

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of the linear constants involved in the
curve, the trend

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will stay the same even for more complex
signal and schemes.

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So the lesson that we learned from the
simple example is that in order

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to reduce the probability of error, we

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should increase G, the amplitude of the
signal.

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But of course,

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increasing G also increases the power of
the transmitted signal, and

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we know that we cannot go above the
channel's power constraint.

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And so that's how the power constraint
limits the reliability of transmission.

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The bi-level signal is keen, is very
instructive, but it's also very

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limited in the sense that we're sending
just one bit per output symbol.

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So to increase the throughput, to increase
the number

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of bits per second that we send over a
channel,

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we can use multilevel signaling.

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There are very many ways to do so, we'll
just look at a few.

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But the fundamental idea is that we take
now, larger chunks of

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bits, and therefore we have alpha bits
that have a higher cardinality.

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So more values in the alpha bit means more

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bits per symbol and therefore a higher
data rate.

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But, not to give the ending away.

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We will see that the power of the signal
will also be dependent on the size of the

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alphabet and so in order not exceed in the
probability of error, given the

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channels power of constraint, we will not
be able to grow the alphabet indefinitely.

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But we can be smart in a way we build this
alphabet.

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And so we will look in some examples.

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The first example is PAM, Pulse-amplitude
Modulation.

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We split the incoming bitstream into
chunks of M bits.

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So that the each chunk corresponds to an
integer between 0 and 2 to the m minus 1.

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We can

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call this sequence of integers, k of n and
this sequence is mapped onto a

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sequence of symbols a of n like so There's
a gain factor G, like always.

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And then we use 2 to the M minus 1, odd
integers around 0.

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So for instance, if M is equal to 2, we
have 0, 1, 2,

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and 3 as potential items for k of n.

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And a of n will be either lets assume

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G is equal to 1 will be either minus 3 or
minus or 1 or 3.

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We will see why we used the odd integers
in just a second.

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At the receiver, the slicer will work by
simply associating to the

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received symbol, the closest odd integer,
always taking the gain into account.

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So graphically again,

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PAM for M equal to 2 and G equal to 1 will
look like this.

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Here are the odd integers.

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The distance between two transmitted
points, or

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transmitted symbols, is 2G, right here G

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is equal to 1, but it would be in general,
2 times the gain.

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And using odd integers creates a zero-mean
sequence.

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If we assume that each symbol is equally
probable, which

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is likely, given that we've used a
scrambler in the transmitter,

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then the resulting mean is zero.

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The analysis of the probability of error
for PAM is

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very similar to what we carried out for
bi-level signaling.

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As a matter of fact, bi-level signaling is
simply PAM with m equal to 1.

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The end result is very similar and its an
exponential decaying of the

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ratio between the power of the signal and
the power of the noise.

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The reason why we don't analyze this
further

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is because we have an improvement in
store.

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And the improvement

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is aimed at increasing the throughput,
increasing the numbers of bit per

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symbol that we can send without

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necessarily increasing the probability of
error.

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So here's a wild idea, let's use complex

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numbers and build a complex valued
transmission system.

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This requires certain suspension of
disbelief for the time

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being, but believe me, it will work in the
end.

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The name for this complex valued mapping
scheme

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is QAM, which is an acronym for quadrature

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amplitude modulation.
And it works like so.

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The mapper takes the incoming bitstream
and splits

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it into chunks of M bits, with M even.

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And then it uses half of the bits to
define

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a PAM sequence which we call a of r of n.

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And the remaining m over two bits to
define

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an another independent PAN sequence a i of
n.

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The final symbol sequence is a sequence of
complex numbers

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where the real part is the first PAM
sequence

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and the imaginary part is a second PAM
sequence.

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And of course in front we have again
factor G.

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So the transmission of alphabet A is given
by points

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in the complex plane with odd valued
coordinates around the origins.

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And the receiver just lies through works
by finding the

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symbol in the alphabet which is closest in
Euclidean distance

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to the received symbol.
Let's look at this graphically.

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This is the set of points for QAM
transmission with M equal to two, which

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corresponds to two bi-level PAM signals on
the real axis and on the imaginary axis.

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So that results into four points.

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If we increase the number of bits per
symbol, we set M equal to four.

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That corresponds to two PAM

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signals with two bits each.
Which makes for a constellation.

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This is how these arrangement of points in
the complex plane are called.

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A constellation of four by four points at

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the odd valued coordinates in the complex
plane.

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If we increase M to 8, then we have

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a 256 point constellation with 16 points
per side.

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Lets look at what happens

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when a symbol is received and how we

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derive an expression from the probability
of the error.

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00:13:13,790 --> 00:13:16,828
If this is the nominal constellation the
transmitter will

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choose one of these values for
transmission, say this one.

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And this value will be corrupted by

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noise in the transmission and the
receiving process.

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And will appear somewhere in the complex
plane,

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00:13:27,420 --> 00:13:31,050
not necessarily exactly on the point it
originates from.

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00:13:32,050 --> 00:13:33,910
The way the slicer operates

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00:13:33,910 --> 00:13:38,490
is by defining decision regions around
each point in the constellation.

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00:13:38,490 --> 00:13:41,730
So suppose for this point here, the
transmitted point, the decision

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region is a square of side, 2G, centered
around the transmitted point.

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00:13:48,680 --> 00:13:51,540
So what happens, is that when we receive

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symbols, they will not fall on the
original

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00:13:54,150 --> 00:13:56,230
point, but as long as they fall within

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00:13:56,230 --> 00:13:58,750
the decision region, they will be decoded
correctly.

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So for instance, here.
We will decode this correctly.

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00:14:01,860 --> 00:14:04,780
Here we will decode this correctly, same
here.

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But this point for instance falls outside
of the decision region

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and therefore it will be associated to a
different constellation point.

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Thereby, causing an error.

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To quantify the probability of error, we
assume as per usual that each

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received symbol is the sum of the
transmitted symbol plus a noise sample.

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Inter of n.

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00:14:23,740 --> 00:14:29,020
And we further assume that this noise is a
complex value

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00:14:29,020 --> 00:14:33,330
Gaussian noise of equal variance in the
complex and real components.

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We're working on a completely digital

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system that operates with complex valued
quantities.

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So we're making a new model for the noise.

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And we will see later how to translate

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the physical real noise into a complex
variable.

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With this assumptions.

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The probability of error is equal to the
probability

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00:14:54,420 --> 00:14:56,680
that the real part of the noise is larger
than

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00:14:56,680 --> 00:15:00,060
G in magnitude, plus the probability that
the imaginary

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00:15:00,060 --> 00:15:03,470
part of the noise is larger than G in
magnitude.

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00:15:03,470 --> 00:15:06,200
We assume that real and imaginary
components of the noise are

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independent, and that's why we can split
the probability like so.

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Now, if you remember the shape of

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the decision region, this condition is
equivalent to

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00:15:15,105 --> 00:15:20,010
saying that the noise is pushing the real
part of the point

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outside of the decision region in either
direction and same for the majority part.

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Now if we develop this, this is to equal 1
minus the probability the real part of

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the noise is less than G and the imaginary
part of the noise is less than G.

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This is the complimentary condition to
what we just wrote above.

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And so this is equal to 1

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minus the integral over the decision
region, d, of

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the complex valued probability density
function for the noise.

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00:15:47,370 --> 00:15:50,610
In order to compute this integral, we're
going to approximate

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00:15:50,610 --> 00:15:55,110
the shape of the decision region with the
inbound circle.

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So instead of using the square.

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00:15:56,735 --> 00:15:59,840
We're going to a circle centered around
the transmission point.

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00:15:59,840 --> 00:16:04,620
When the conciliation is very dense, this
approximation is quite accurate.

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00:16:04,620 --> 00:16:05,550
With this approximation,

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00:16:05,550 --> 00:16:09,320
we can compute the integral exactly for a
Gaussian distribution.

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00:16:09,320 --> 00:16:12,170
And if we assume that the variance of the
noise is

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00:16:12,170 --> 00:16:17,920
sigma 0 squared over 2 in each component,
real and imaginary.

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00:16:17,920 --> 00:16:19,980
It turns out that the probability of error
is equal

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00:16:19,980 --> 00:16:23,540
to e to the minus g square over sigma 0
square.

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00:16:23,540 --> 00:16:26,360
Now to obtain a probability of error as a
function of the signal

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to noise ratio, we have to compute the
power of the transmitted signal.

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00:16:31,170 --> 00:16:34,550
So if all symbols are equiprobable and
independent.

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00:16:34,550 --> 00:16:37,180
It turns out that the variance of the
signal is G

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00:16:37,180 --> 00:16:40,410
square times 1 over 2 to the power of M,
which

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00:16:40,410 --> 00:16:43,840
is a probability of each symbol times the
sum of over

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00:16:43,840 --> 00:16:47,632
all symbols in the alphabet of the
magnitude of the symbol squared.

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00:16:47,632 --> 00:16:53,670
Now it's a little bit tedious but we can
solve it exactly for M

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00:16:53,670 --> 00:16:56,280
and it turns out that the power of the of
the transmitted signal is G

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00:16:56,280 --> 00:16:59,520
square, 2 3rds, 2 the M minus 1.

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00:16:59,520 --> 00:17:02,960
Now if we plug this in to the formula for
the probability of error, that we've

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00:17:02,960 --> 00:17:07,040
seen before, we get that the result is

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00:17:07,040 --> 00:17:11,400
an exponential function where the argument
is minus 3.

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00:17:11,400 --> 00:17:16,490
That multiplies 2 to the minus m plus 1,
that multiplies the signal's noise ratio.

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00:17:17,600 --> 00:17:21,490
We can plot this probability of error in a
log log scale, like we did before.

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And we can parameterize the curve as a

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00:17:24,350 --> 00:17:27,570
function of the number of points in the
constellation.

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So here you have the curve for a four
point constellation.

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00:17:30,880 --> 00:17:34,680
Here's the curve for 16 points.
And here's the curve for 64 points.

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00:17:34,680 --> 00:17:37,150
Now, you can see that for a given signal
to noise

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00:17:37,150 --> 00:17:41,450
ratio, the probability of error increases
with the number of points.

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00:17:41,450 --> 00:17:42,630
Why is that?

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00:17:42,630 --> 00:17:44,820
Well, if the signal to noise remains the
same,

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00:17:44,820 --> 00:17:46,510
and we assume that the noise is always at
the

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00:17:46,510 --> 00:17:47,505
same level.

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00:17:47,505 --> 00:17:50,690
Then it means that the power of the signal
remains constant as well.

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00:17:50,690 --> 00:17:54,710
In that case, if the number of points
increases G has to become

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00:17:54,710 --> 00:17:59,490
smaller in order to accommodate a larger
number of points for the same power.

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00:17:59,490 --> 00:18:03,490
But if G becomes smaller then the decision
reaches become smaller.

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00:18:03,490 --> 00:18:06,010
The separation between points becomes
smaller and

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00:18:06,010 --> 00:18:08,740
the decision process becomes more
vulnerable to noise.

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00:18:10,300 --> 00:18:11,980
So in the end, here is the final recipe

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00:18:11,980 --> 00:18:14,150
to design a QAM transmitter.

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00:18:14,150 --> 00:18:16,628
First you pick a probability of error that
you can live with.

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00:18:16,628 --> 00:18:18,380
In general 10 to the minus 6 is

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00:18:18,380 --> 00:18:20,930
an acceptable probability of error at the
symbol level.

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00:18:22,070 --> 00:18:23,720
Then you find out the signal to noise

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00:18:23,720 --> 00:18:26,550
ratio that is imposed by the channel's
power constraint.

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00:18:26,550 --> 00:18:32,140
Once you have that, you can find the size
of your constellation by finding m which

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00:18:32,140 --> 00:18:37,530
based on the previous equations is the log
in base 2 of 1 minus 3 over 2

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00:18:37,530 --> 00:18:39,890
times the signal to noise ratio divided by

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00:18:39,890 --> 00:18:42,350
the natural logarithm of the probability
of error.

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00:18:42,350 --> 00:18:45,560
Of course you will have to round this to a
suitable integer value and

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00:18:45,560 --> 00:18:49,170
potentially to an even power of 2 in order
to have the square constellation.

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00:18:49,170 --> 00:18:53,280
The final data rate of your system will be
m the number of bits

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00:18:53,280 --> 00:18:58,460
per symbol, times w which if you remember
is the board rate of system.

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00:18:58,460 --> 00:19:02,990
And corresponds to the bandwidth allowed
for by the channel.

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00:19:02,990 --> 00:19:07,820
So, we know how to fit the bandwidth
constraint by upsampling.

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00:19:07,820 --> 00:19:12,560
With QAM, we know how many bits per symbol
we can use given the power constraint.

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00:19:12,560 --> 00:19:14,310
And so we know the theoretical throughput

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00:19:14,310 --> 00:19:16,720
of the transmitter, for a given
reliability figure.

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00:19:18,370 --> 00:19:20,560
However, the question remains, how are we
going

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00:19:20,560 --> 00:19:24,910
to send complex value symbols over a
physical channel.

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00:19:24,910 --> 00:19:28,880
It's time therefore to stop the suspension
of disbelief and

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00:19:28,880 --> 00:19:33,080
look at techniques to do complex signaling
over a real value channel.

