I was taking a look at Microsoft Quantum Development Kit Documentation when I've stumbled on the Teleportation technique tutorial. It says:
1. Entangling here and there by applying a Hadamard gate and CNOT gate.
2. Sending the message using a CNOT gate and a Hadamard gate.
3. Taking a measurement of the first and second qubits, message and here.
4. Applying a NOT gate or a Z gate, depending on the result of the measurement in step 3.
I understand that Hadamarding a qubit and then using CNOT on it and on other qubit entangles them and that's an important part of Teleportation. What I don't understand is how using CNOT on the message qubit and then Hadamarding it would help at all on the Teleportation process.
Why would I want my message qubit to enter in a superposition state? I don't want to get a random value out of my message, instead, I want it to be transmitted to another qubit.
I get that if a qubit A is collapsed to a message that I'd like to send and if a qubit B is initialized to |0>, CNOT(A, B) would transmit the message. However, what is the purpose of doing this operation if B is in a 50-50 superposition state? I mean, B's wave function would change from sqrt(1/2)|0> + sqrt(1/2)|1> to sqrt(1/2)|1> + sqrt(1/2)|0> How would that help in any way?
But the worst part for me is that the method above works just perfectly. There must be something very basic I'm missing. Thanks for reading.
The best way to understand simple quantum algorithms is to follow the math.
Applying a CNOT to the message qubit and Alice's part of the Bell pair entangles all three qubits, so that the information from the message qubit is distributed across the whole system. Applying a Hadamard gate to the former message qubit and then measuring it and the Alice's qubit makes sure that all the information is concentrated in Bob's qubit and the classical measurement results. But you have to do the math to see what are the states of the system between those operations and why do you have to apply the fixups that the algorithm prescribes.
I will not go into the math here (mostly because it's a huge pain to do so without TeX) - the link you gave does all the math in great detail. To augment the formulas, you can look for an interactive demo which shows the process of teleporting a specific state, such as this one.
One way to gain intuition is to not think about those gates as a processing step, but rater a measurement in the Bell basis. (That's how you measure two qubits in the bel basis CNOT + H on qubit 1 + measure)
If you think about it like that then the process is this: you take the qubit to be transmitted and measure it in the Bell basis with qubit A - "you measure the correlation between that qubit and qubit A". Then you take the "measured correlation" and apply "inversely" it to qubit B (which is entangled with A) giving you the original qubit again.
Related
So, I have a state of 3 qubits that is in one of the states in the picture. How can I find out which state it is in?
I tried to measure the qubits but the amplitudes are 1/3 for each, so...
This is task 1.15 from the Measurements kata, so I'll outline the solution broadly and point you to the workbook for that task for the formulas and details that are painful to spell out on StackOverflow without LaTeX support.
When you need to distinguish two orthogonal quantum states, you can first apply a unitary to them to rotate them to two different orthogonal quantum states that are easy to distinguish - for example, states that have different basis states in their superposition makeup.
In this case,
We can first apply some rotation gates (R1 gate in Q# or similar gates) to the second and the third qubits to get rid of the đťś” amplitudes in the first state, converting it into the W state.
Then, apply adjoint of the transformation you'd use to prepare the W state from the |000âź© state, so that this state ends up becoming the |000âź© state.
Since the transformations you've applied are unitaries and they preserve the product of state vectors, you know that the vectors remain orthogonal after their application. This means that if you do the measurements now, you'll always get 000 for the first state, and some other basis state for the second one.
I would like to make a quantum circuit from the following matrix.
matrix to be transformed into qubit operations
How can I decompose this matrix into qubit operations such as Rotation Y, Control-NOT and so on ?
FYI, I read a book named "Quantum Computation and Quantum Information" written by Nielsen & Chuang, in particular Section 4.5.
A universal method to decompose 2-qubit unitaries into primitive gates is sometimes referred to as "Krauss-Cirac decomposition". Here are several sources:
Optimal Quantum Circuits for General Two-Qubit Gates by Vatan and Williams,
Optimal Creation of Entanglement Using a Two–Qubit Gate by Kraus and Cirac.
“Explorations in Quantum Computing” by Williams, chapter 2.
As a side note, such questions are usually better received at Quantum Computing StackExchange.
I would like to get familiar with quantum computing basics.
A good way to get familiar with it would be writing very basic virtual quantum computer machines.
From what I can understand of it, the, effort of implementing a single qubit cannot simply be duplicated to implement a two qubit system. But I don't know how I would implement a single qubit either.
How do I implement a qubit?
How do I implement a set of qubits?
Example Code
If you want to start from something simple but working, you can play around with this basic quantum circuit simulator on jsfiddle (about ~2k lines, but most of that is UI stuff [drawing and clicking] and maths stuff [defining complex numbers and matrices]).
State
The state of a quantum computer is a set of complex weights, called amplitudes. There's one amplitude for each possible classical state. In the case of qubits, the classical states are just the various states a normal bit can be in.
For example, if you have three bits, then you need a complex weight for the 000, 001, 010, 011, 100, 101, 110, and 111 states.
var threeQubitState = new Complex[8];
The amplitudes must satisfy a constraint: if you add up their squared magnitudes, the result is 1. Classical states correspond to one amplitude having magnitude 1 while the others are all 0:
threeQubitState[3] = 1; // the system is 100% in the 011 state
Operations
Operations on quantum states let you redistribute the amplitude by flowing it between the classical states, but the flows you choose must preserve the squared-magnitudes-add-up-to-1 property in all cases. More technically, the operation must correspond to some unitary matrix.
var myOperation = state => new[] {
(state[1] + state[0])/sqrt(2),
(state[1] - state[0])/sqrt(2),
state[2],
state[3],
state[4],
state[5],
state[6],
state[7]
};
var myNewState = myOperation(threeQubitState);
... and those are the basics. The state is a list of complex numbers with unit 2-norm, the operations are unitary matrices, and the probability of measuring a state is just its squared amplitude.
Etc
Other things you probably need to consider:
What kinds of operations do you want to include?
A 1-qubit operation is a 2x2 matrix and a 3-qubit operation is an 8x8 matrix. How do you convert a 1-qubit operation into an 8x8 matrix when applying it to a single qubit in a 3-qubit state? (Use the Kronecker Product.)
What kinds of tricks can you use to speed up the simulation? For example, if only a few states are non-zero, or if the qubits are not entangled, there's no need to do a full matrix multiplication.
How does the user tell the simulation what to do? How can you represent what's going on for the user? There's an awful lot of numbers flowing around...
I don't actually know the answer, but an interesting place to start reading about qubits is this article. It doesn't describe in detail how entangled qubits work, but it hints at the complexity involved:
If this is how complicated things can get with only two qubits, how
complicated will it get for 3 or 4, or 100? It turns out that the
state of an N-qubit quantum computer can only be completely defined
when plotted as a point in a space with (4^N-1) dimensions. That means
we need 4^N good old fashion classical numbers to simulate it.
Note that this is the maximum space complexity, which for example is about 1 billion numbers (2^30=4^15) for 15 qubits. It says nothing about the time complexity of a simulation.
The article that #Qwertie cites is a very good introduction. If you want to implement these on your computer, you can play with the libquantum simulator, which implements sophisticated quantum operations in a C library. You can look at this example to see what using the code is like.
The information is actually stored in the interaction between different Qbits, so no implementing 1 Qbit will not translate to using multiple. I'd think another way you could play around is to use existing languages like QCL or google QCP http://qcplayground.withgoogle.com/#/home to play around
What is a simple way to see if my low-pass filter is working? I'm in the process of designing a low-pass filter and would like to run tests on it in a relatively straight forward manner.
Presently I open up a WAV file and stick all the samples in a array of ints. I then run the array through the low-pass filter to create a new array. What would an easy way to check if the low-pass filter worked?
All of this is done in C.
You can use a broadband signal such as white noise to measure the frequency response:
generate white noise input signal
pass white noise signal through filter
take FFT of output from filter
compute log magnitude of FFT
plot log magnitude
Rather than coding this all up you can just dump the output from the filter to a text file and then do the analysis in e.g. MATLAB or Octave (hint: use periodogram).
Depends on what you want to test. I'm not a DSP expert, but I know there are different things one could measure about your filter (if that's what you mean by testing).
If the filter is linear then all information of the filter can be found in the impulse response. Read about it here: http://en.wikipedia.org/wiki/Linear_filter
E.g. if you take the Fourier transform of the impulse response, you'll get the frequency response. The frequency response easily tells you if the low-pass filter is worth it's name.
Maybe I underestimate your knowledge about DSP, but I recommend you to read the book on this website: http://www.dspguide.com. It's a very accessible book without difficult math. It's available as a real book, but you can also read it online for free.
EDIT: After reading it I'm convinced that every programmer that ever touches an ADC should definitely have read this book first. I discovered that I did a lot of things the difficult way in past projects that I could have done a thousand times better when I had a little bit more knowledge about DSP. Most of the times an unexperienced programmer is doing DSP without knowing it.
Create two monotone signals, one of a low frequency and one of a high frequency. Then run your filter on the two. If it works, then the low frequency signal should be unmodified whereas the high frequency signal will be filtered out.
Like Bart above mentioned.
If it's LTI system, I would insert impulse and record the samples and perform FFT using matlab and plot magnitude.
You ask why?
In time domain, you have to convolute the input x(t) with the impulse response d(t) to get the transfer function which is tedious.
y(t) = x(t) * d(t)
In frequency domain, convolution becomes simple multiplication.
y(s) = x(s) x d(s)
So, transfer function is y(s)/x(s) = d(s).
That's the reason you take FFT of impulse response to see the behavior of the filter.
You should be able to programmatically generate tones (sine waves) of various frequencies, stuff them into the input array, and then compare the signal energy by summing the squared values of the arrays (and dividing by the length, though that's mathematically not necessary here because the signals should be the same length). The ratio of the output energy to the input energy gives you the filter gain. If your LPF is working correctly, the gain should be close to 1 for low frequencies, close to 0.5 at the bandwidth frequency, and close to zero for high frequencies.
A note: There are various (but essentially the same in spirit) definitions of "bandwidth" and "gain". The method I've suggested should be relatively insensitive to the transient response of the filter because it's essentially averaging the intensity of the signal, though you could improve it by ignoring the first T samples of the input, where T is related to the filter bandwidth. Either way, make sure that the signals are long compared to the inverse of the filter bandwidth.
When I check a digital filter, I compute the magnitude response graph for the filter and plot it. I then generate a linear sweeping sine wave in code or using Audacity, and pass the sweeping sine wave through the filter (taking into account that things might get louder, so the sine wave is quiet enough not to clip) . A visual check is usually enough to assert that the filter is doing what I think it should. If you don't know how to compute the magnitude response I suspect there are tools out there that will compute it for you.
Depending on how certain you want to be, you don't even have to do that. You can just process the linear sweep and see that it attenuated the higher frequencies.
I have built my first neural network in python, and i've been playing around with a few datasets; it's going well so far !
I have a quick question regarding modelling events with multiple outcomes: -
Say i wish to train a network to tell me the probability of each runner winning a 100m sprint. I would give the network all of the relevant data regarding each runner, and the number of outputs would be equal to the number of runners in the race.
My question is, using a sigmoid function, how can i ensure the sum of the outputs will be equal to 1.0 ? Will the network naturally learn to do this, or will i have to somehow make this happen explicitly ? If so, how would i go about doing this ?
Many Thanks.
The output from your neural network will approach 1. I don't think it will actually get to 1.
You actually don't need to see which output is equal to 1. Once you've trained your network up to a specific error level, when you present the inputs, just look for the maximum output in your output later. For example, let's say your output layer presents the following output: [0.0001, 0.00023, 0.0041, 0.99999412, 0.0012, 0.0002], then the runner that won the race is runner number 4.
So yes, your network will "learn" to produce 1, but it won't exactly be 1. This is why you train to within a certain error rate. I recently created a neural network to recognize handwritten digits, and this is the method that I used. In my output layer, I have a vector with 10 components. The first component represents 0, and the last component represents 9. So when I present a 4 to the network, I expect the output vector to look like [0, 0, 0, 0, 1, 0, 0, 0, 0, 0]. Of course, it's not what I get exactly, but it's what I train the network to provide. So to find which digit it is, I simply check to see which component has the highest output or score.
Now in your second question, I believe you're asking how the network would learn to provide the correct answer? To do this, you need to provide your network with some training data and train it until the output is under a certain error threshold. So what you need is a set of data that contains the inputs and the correct output. Initially your neural network will be set up with random weights (there are some algorithms that help you select better weights to minimize training time, but that's a little more advanced). Next you need a way to tell the neural network to learn from the data provided. So basically you give the data to the neural network and it provides an output, which is highly likely to be wrong. Then you compare that data with the expected (correct) output and you tell the neural network to update its weights so that it gets closer to the correct answer. You do this over and over again until the error is below a certain threshold.
The easiest way to do this is to implement the stochastic backpropagation algorithm. In this algorithm, you calculate the error between the actual output of the neural network and the expected output. Then you backpropagate the error from the output layer all the way up to the weights to the hidden layer, adjusting the weights as you go. Then you repeat this process until the error that you calculate is below a certain threshold. So during each step, you're getting closer and closer towards your solution.
You can use the algorithm described here. There is a decent amount of math involved, so be prepared for that! If you want to see an example of an implementation of this algorithm, you can take a look at this Java code that I have on github. The code uses momentum and a simple form of simulated annealing as well, but the standard backpropagation algorithm should be easily discernible. The Wikipedia article on backpropagation has a link to an implementation of the backpropagation algorithm in Python.
You're probably not going to understand the algorithm immediately; expect to spend some time understanding it and working through some of the math. I sat down with a pencil and paper as I was coding, and that's how I eventually understood what was going on.
Here are a few resources that should help you understand backpropagation a little better:
The learning process: backpropagation
Error backpropagation
If you want some more resources, you can also take a look at my answer here.
Basically you want a function of multiple real numbers that converts those real numbers into probabilities (each between 0 to 1, sum to 1). You can this easily by post processing the output of your network.
Your network gives you real numbers r1, r2, ..., rn that increases as the probability of each runner wins the race.
Then compute exp(r1), exp(r2), ..., and sum them up for ers = exp(r1) + exp(r2) + ... + exp(rn). Then the probability that the first racer wins is exp(r1) / ers.
This is a one use of the Boltzman distribution. http://en.wikipedia.org/wiki/Boltzmann_distribution
Your network should work around that and learn it naturally eventually.
To make the network learn that a little faster, here's what springs to mind first:
add an additional output called 'sum' (summing all the other output neurons) -- if you want all the output neurons to be in an separate layer, just add a layer of outputs, first numRunners outputs just connect to corresponding neuron in the previous layer, and the last numRunners+1-th neuron you connect to all the neurons from the previous layer, and fix the weights to 1)
the training set would contain 0-1 vectors for each runner (did-did not run), and the "expected" result would be a 0-1 vector 00..00001000..01 first 1 marking the runner that won the race, last 1 marking the "sum" of "probabilities"
for the unknown races, the network would try to predict which runner would win. Since the outputs have contiguous values (more-or-less :D) they can be read as "the certainty of the network that the runner would win the race" -- which is what you're looking for
Even without the additional sum neuron, this is the rough description of the way the training data should be arranged.