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Hi and welcome to Module 9.6 of Digital 
Signal Processing. 

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This is our last module in our Data 
Communications section, and with 

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everything that we've learned so far we 
will be able to look inside an ADSL modem 

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and see how that works. 
We will start by examining the nature of 

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the channel. 
That ABSL works on. 

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Namely the copper wire links your home to 
the nearest central office. 

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We will look at the signaling strategy 
that is best put on place on such a 

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channel. 
And finally we will look at a very 

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efficient implementation of that 
signalling strategy that goes under the 

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name of discrete multi-tone modulation. 
If we just take an abstract view of the 

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telephone network today, we see that we 
have a link that goes. 

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From the home to the central office, and 
in the central office, a fundamental 

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split takes place. 
The voice communication, when you talk on 

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the phone, is sent to the voice network 
and then relayed to the, what is called 

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the plain old telephone system, POTS. 
The data part of your communication when 

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you use the ADSL is separated from the 
voice content and sent to a DSLAM. 

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DSLAM stands for digital subscriber line 
access multiplier. 

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And it's fundamentally a bank of modems 
that manage to handle multiple 

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communications at the same time. 
And the data here then goes on to the 

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internet in digital format. 
So, if you want, what we're really 

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interested in is how to send the data 
from your home to the central office, 

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because what happens afterwards is 
already entirely in the digital domain. 

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But here, we have what is called a last 
mile, a piece of copper wire, namely an 

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analog channel. 
That connects your home to the central 

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office. 
Now a copper wire has, naturally, a very 

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large bandwidth in excess of one 1MHz. 
But because of the width of the bandwidth 

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and because the wire is not shielded, it 
is actually likely to pick up a lot of 

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interference and noise. 
If we look at how the ADSL channel is 

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organized and we're showing here just the 
positive frequencies. 

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We can see three distinct regions. 
The first one is the part reserved to the 

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telephone conversation, this is the base 
band part of the channel up to about four 

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kilohertz. 
Then we have a region that is devoted to 

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the upstream part of the data 
communication, the data that you send up 

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to the internet, and then a downstream 
part that is much larger that is used for 

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data download. 
This asymmetry between upstream and 

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downstream is actually the reason why the 
communication protocol is called ADSL. 

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ADSL stands for Asymmetric Digital 
Subscriber Line. 

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If we now look at all the nasty things 
that can happen on the channel when we 

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send data. 
We can identify three fundamental sources 

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of worry. 
The first one is an attenuation curve for 

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the channel that is completely uneven. 
This could be due to imperfection in the 

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wire, parasitic capacitance, and so on, 
so forth. 

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Then we might have very large noise or 
interference in certain regions of the 

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spectrum. 
For instance, you turn on your vacuum 

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cleaner, and that raises the noise floor 
in the certain frequency band. 

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And thirdly we have very localized 
interference from radio communications 

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now the radio band starts well within the 
bandwidth of the ADSL channel for 

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instance from 15 to a hundred kilohertz 
here you have ship to shore 

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communication. 
Up to 500 kilohertz, you have airplane 

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communication, and over 500 kilohertz you 
have the AM radio band. 

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So if you live near a radio station, for 
instance, tough luck. 

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You have a lot of interference in the 
upper regions of the ADSL band. 

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Since the channel is so wide, and the 
type of disturbances is so diverse, it 

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would be extraordinarily difficult to try 
to equalize and compensate for this 

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problems on a global scale. 
So the ideal, instead, is to divide the 

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channel into independent sub channels. 
And create each sub channel separately. 

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So here for instance, this channel 
contains, highly localized, radio 

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frequency interference would not probably 
be used because it would be to difficult 

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to compensate for that. 
Here on these channels the noise level is 

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different and so for instance we can use 
different signalling strategy according 

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to the local signal to noise ratio. 
And similarly here for channels that have 

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a very large attenuation, probably 
wouldn't be worth while to try and send 

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data over these thins. 
But on the other hand that we will try to 

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exploit the cleanest sub channels to send 
maximum amount of data. 

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Now to formalize the sub channel 
structure. 

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Suppose that we want to allocate N 
subchannels over the total positive 

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bandwidth. 
We want the subchannels to have equal 

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bandwidth, so their bandwidth will be F 
max over N, where F max is the maximum 

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frequency allowed for by the channel. 
And we equally space the subchannels, by 

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sensoring them over k F max over N, with 
k that goes from 0 to big N minus 1. 

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This means that the first channel K equal 
to 0, will be bass-band. 

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And then the subsequent channels will be 
bass-band, with center frequencies given 

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by this formula. 
Now we want to translate this design, to 

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the digital domain, so we pick a sampling 
frequency that is at least twice the 

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maximum frequency in the channel. 
But careful now, because Fmax is quite 

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high. 
The center frequency for each sub channel 

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will be omega K equal to 2 pi kFmax over 
N divided by the sampling frequency. 

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And if we sample at the[UNKNOWN] 
frequency, so Fs is equal twice Fmax, 

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then omegak becomes simple 2 pi over 2N 
times k. 

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We will not simplify the 2's in the 
fraction, because they will be useful 

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later. 
The bandwidth of each subchannel, is also 

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2 pi over 2N. 
And so, if we want to send symbols over 

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any of these subchannels, remember the 
modulation scheme that we've seen in the 

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previous modules, then we will have to 
use an up-sampling factor, k, that is at 

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least 2N. 
If we plot the result in visual domain. 

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Let's suppose that we just want to have 
three sub-channels, we have something 

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that looks like this. 
The center frequencies will be multiples 

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of 2 pi over 6, so we'll have 0. 
2 pi over 6 and 4 pi over 6 and we will 

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center channels over these frequencies 
and the band width of each channel will 

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be 2 pi over 6. 
So the 1st channel is the base band 

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channel and then we have two pass band 
channels with of course their negative 

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frequency counterpart. 
The next step in ADSL communication is to 

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put a QAM modem on each subchannel 
independently and we will decide on the 

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data rate for each modem. 
Based on the signal to noise ratio of 

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each sub channel. 
So if the noise floor is low, then we 

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will have a large constellation for that 
sub channel and vice versa. 

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On channels that are unusable because of 
noise or interference, we will just send 

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zeroes and we will not care about that. 
The structure of the SQL scheme is of 

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course going to be communicated From the 
transmitter to the receiver so that the 

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receiver knows where to expect data. 
This is part of the handshaking procedure 

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between transmitter and receiver. 
Now let's look more in detail at the 

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structure of the modem that we use on 
each sub-channel. 

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This is a classic modulation scheme where 
we start with a sequence of symbols as 

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produced by the mapper. 
Then we have an up-sampling by a factor 

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of 2 N. 
So inserting two n minus one zero's every 

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other sample and then filtering the 
sequence with a low pass, usually a 

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raised cosine, with a cutoff frequency 
two pi over two n, in this case. 

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This produces the complex base band 
signal bk of n and this. 

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Complex baseband signal gets modulated 
with a complex exponential whose 

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frequency is indexed by the channel 
number. 

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And this is the center frequency of each 
channel. 

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Omega ck is equal to 2pi over 2n times k. 
And here we have, finally, the pass band 

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signal. 
That fits the prescribed bandwidth of the 

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kth channel. 
And normally, if we just had one channel, 

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we would put here a block that computes 
the real part and then our d2a converter. 

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But here we have several modems in 
parallel. 

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And so we have a structure that looks 
like this. 

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Each channel will have two things that 
vary with respect to the others: the 

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frequency of the modulation and of 
course, the series of symbols produced by 

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the mapper. 
We sum all the complex base band signals 

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together before taking the real part, and 
then send in the signal To the D to A 

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converter. 
Now this picture should ring a bell. 

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An indeed we have seen something that was 
very very close to this back in module 

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4.3. 
So here's the picture to jog your memory 

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and remember that DFT reconstruction 
formula could be interpreted as. 

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A bank of n oscillators. 
Each oscillator would operate at a 

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frequency that was two pi over n times k. 
And we would scale each oscillator with 

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an amplitude, a of k, and with a phase 
offset, five k. 

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We would run this machine for big n 
samples. 

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And we will get our signal out. 
Now, the difference between this scheme 

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and what we just saw is fundamentally 
that, in this scheme, a of k and phi of k 

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are kept constant for the whole duration 
of the generation process. 

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So while n goes from zero to big n minus 
one, a of k and phi of k stay the same. 

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Whereas in the modem scheme that we seen 
before. 

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The symbol sequence which is a complex 
symbol sequence so embeds both magnitude 

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and phase will change at each new value 
of n. 

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So is there a way to map the modem 
structure to the inverse DFT structure? 

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We can do that if we manage to find a way 
to keep the symbols constant over the 

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whole duration. 
So we will show how to do that and we 

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will show that if we manage to do that, 
then ADSL transmission can be efficiently 

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implemented with simple an inverse FFT. 
The name of this technique is discrete 

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multitone modulation. 
So the great ADSL trick is very simple. 

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Instead of using a[INAUDIBLE] sign in the 
up sampler, let's use a bad filter. 

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Simply the indicator function for the 
interval 0 to 2n-1 and see what happens. 

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So the impulse response of the upsampling 
filter is now this one. 

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And please notice that this is just an 
un-normalized moving average filter and 

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the frequency response of course is this 
one, we have seen it many times before. 

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With the first 0 here in pi over N. 
If we compare the frequency response of 

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the moving average if you want, or the 
indicator function with that of the 

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filter that we should be using, namely a 
low pass filter with cut of pi over 2N. 

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Then we see that the performance of the 
filter is not very good. 

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Nonetheless, the thing will work, 
especially thanks to some clever little 

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tricks in the way we choose the 
transmission symbols. 

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But we will not have time to go into 
that. 

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So let's go back to the subchannel modem. 
The thing to remark here is that the 

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symbols from the mapper. 
Come in at a rate of B symbols per 

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second. 
And because of the upsampling, samples 

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out of the modulator, come out at a rate 
of 2NB samples per second. 

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So this part works much faster than this 
part here. 

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Now the carrier, is periodic with period 
2N. 

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And so, each symbol Will influence a full 
period of the carrier. 

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If we use a standard low pass filter 
here, for every value of n here, so for 

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every value of the carrier there will be 
a different value in this Base band 

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sequence that comes out of the sample. 
On the other hand if we use the indicator 

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function as the input response the net 
result is that the values of bk of n will 

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be constant over chunks of 2N samples. 
In that case we can simplify this whole 

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scheme like so where now the only clock 
in the system is the output clock N. 

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So now the oscillator in the modulator. 
Runs freely, at a frequency which is a 

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multiple of 2 pi over 2N. 
This frequency is periodic, with period 

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2N, so. 
And for each chunk of 2N samples, we go 

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look for the symbol. 
The corresponds to this interval so if 

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say n is equal to 0 here, n is equal to 
2n here, n is equal to 4n here and n is 

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equal to 6n here for this interval we 
will go look for a of zero and multiply 

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this portion of the carrier by this 
value. 

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Able to look for a one, here a two, and 
so on and so on. 

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So with this simplification, the whole 
transmitter can be sketched like so. 

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We have the symbols from different sub 
channels that get multiplied by the 

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carrier and kept constant over intervals 
of two and outward samples. 

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The whole thing gets summed together. 
We get the aggregate bandpass signal, we 

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take the real part, and we're ready for 
the D2A converter. 

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We can now write explicitly, the formula 
for the aggregate bandpass signal c n. 

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And this is the sum over all subchannels, 
of the symbol For that subchannel for 

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that interval, multiplied by e to the j 
two pi over two n, nk. 

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Now because of the way the index to the 
ak sequence is computed, these symbols 

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will stay constant over intervals for the 
output index. 

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That are 2n long. 
So for instance, for small n that goes 

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from 0, to 2n minus 1, these values will 
stay constant. 

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And again, for values of the output index 
that g from 2n. 

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2, 4 and minus 1. 
So we could compute two big N values for 

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the sequence CN in one fell swoop if we 
exploit the fact that this guy looks 

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remarkably like an inverse DFT. 
As a matter of fact, by looking at the 

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argument here, we can say that this is 
almost an inverse DFT over two big end 

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points. 
The two things that are missing are the 

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normalizing factor in front, 1 over 2N 
And the terms in the sum for the index k 

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that goes from n to 2n minus 1. 
But that's not a problem. 

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We can supplement this elements. 
And so we can compute a chunk of two big 

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n output samples in one go as an inverse 
dft over two n points of a vector that is 

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given by n channel symbols. 
And has another big n zeros appended to 

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the end of it. 
The index for the subchannel symbols is 

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given by the value of the output index 
divided by 2 n and we take the integer 

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part. 
But we can do even better because in the 

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end, remember, we're interested in the 
real part of the vector c of n. 

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And we can write that real part as c of n 
plus the conjugate of c of n divided by 

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2. 
Now it is easy to prove, and it's left as 

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an exercise, that the conjugate of the 
inverse DFT of a vector, is equal to the 

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inverse DFT of the conjugate of the time 
reversed vector. 

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And, when we time reverse a finite length 
vector It's useful to think of the 

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periodic extension. 
With this result and knowing that c of n 

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00:15:47,658 --> 00:15:55,390
is equal to 2n times the inverse DFT of a 
vector that is zeros in its latter part. 

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We can sum cn with it's conjugate to 
obtain that the real part of cn is equal 

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to N times the inverse DFT of a vector 
that is given by twice the symbol for the 

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base band sub channel. 
The reason why we can write this is 

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because the baseband signal will always 
have real value symbols, because it's a 

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baseband, followed by the complex symbols 
for the n minus one remaining 

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00:16:22,485 --> 00:16:27,660
subchannels, followed by The conjugate of 
the symbols, for the N minus 1 remaining 

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subchannels, but going from channel N 
minus 1, to channel 1. 

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Schematically, we can draw up the ADSL 
transmitter, as one big inverse FFT, and 

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the inputs to this FFT are twice the 
baseband symbol. 

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Followed by the symbols for the 
subchannels from one to n minus one, and 

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then we take these values, we conjugate 
them, and we flip their order, and we put 

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those in the remaining inputs of the FFT. 
Now we run the inverse FFT and we get two 

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n output samples in one go. 
We use a parrel to serial device to 

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output the samples one at a time and here 
we have our d to i converter to put them 

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on the channel. 
Once the n samples have been put out we 

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go back, we fetch another set of n 
symbols from the n mamppers of the sub 

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channels. 
Then we repeat the process. 

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00:17:26,624 --> 00:17:32,153
An actual ADSL modem uses a maximum 
frequency for the channel of 1004 

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kilohertz divides this channel in to 256 
sub-channels. 

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00:17:38,766 --> 00:17:43,842
Each QAM modem for the sub-channel. 
Can independently choose between zero and 

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fifteen bits per symbol. 
Now the first seven channels are left off 

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because that is the band used by the 
voice communication over a telephone 

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channel. 
Channels seven through 31 are used for 

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data upstream. 
And the rest is left for data downstream 

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for a maximum theoretical throughput of 
14.9 megabits per second. 

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00:18:04,690 --> 00:18:08,274
This would happen if all the downstream 
sub channels could use their maximum 

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00:18:08,274 --> 00:18:11,510
theoretical rate Which is a rare 
occurrence. 

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00:18:11,510 --> 00:18:17,732
And these are the specs of the on-line 
modem that you most probably used to 

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watch this on-line class. 

