I want to make a 8 bit s-box using 4 bit s-box. So is there any easy, understandable algorithm or source to help me?
There is always an easy one, however, that is not guaranteed to carry the security.
4-bit SBox has 4-bit input and 4-bit output. So we can consider that as an array of size 16' S16[16]
8-bit SBox has 8-bit input and 8-bit output. So we can consider that as an array of size 64' S64[64]
Initially, we can copy 4 copies of S16 into S64, however, that won't be a good SBox and the inverse will be failed.
Now, we can use the extra 4-bit to modify the 2,3, and 4 copies as
# copy the 4-Box directly
for i in range(0..16):
S64[i] = S16[i]
# Add 16 the 4-Box's elements then assign
for i in range(16..32):
S64[i] = S16[i-16] *2
# Now add 32 to the first half then assign.
for i in range(32..64):
S64[i] = 32 + S64[i-32] ```
So
the 1st quarter has elements in the range [ 0,15]
the 2nd quarter has elements in the range [16,31]
the 3rd quarter has elements in the range [32,47]
the 4th quarter has elements in the range [48,64]
Related
I have a hardware-based boolean generator that generates either 1 or 0 uniformly. How to use it to make a uniform 8-bit integer generator? I'm currently using the collected booleans to create the binary string for the 8-bit integer. The generated integers aren't uniformly distributed. It follows the distribution explained on this page. Integers with ̶a̶ ̶l̶o̶t̶ ̶o̶f̶ ̶a̶l̶t̶e̶r̶n̶a̶t̶I̶n̶g̶ ̶b̶I̶t̶s̶ the same number of 1's and 0's such as 85 (01010101) and -86 (10101010) have the highest chance to be generated and integers with a lot of repeating bits such as 0 (00000000) and -1 (11111111) have the lowest chance.
Here's the page that I've annotated with probabilities for each possible 4-bit integer. We can see that they're not uniform. 3, 5, 6, -7, -6, and -4 that have the same number of 1's and 0's have ⁶/₁₆ probability while 0 and -1 that all of their bits are the same only have ¹/₁₆ probability.
.
And here's my implementation on Kotlin
Based on your edit, there appears to be a misunderstanding here. By "uniform 4-bit integers", you seem to have the following in mind:
Start at 0.
Generate a random bit. If it's 1, add 1, and otherwise subtract 1.
Repeat step 2 three more times.
Output the resulting number.
Although the random bit generator may generate bits where each outcome is as likely as the other to be randomly generated, and each 4-bit chunk may be just as likely as any other to be randomly generated, the number of bits in each chunk is not uniformly distributed.
What range of integers do you want? Say you're generating 4-bit integers. Do you want a range of [-4, 4], as in the 4-bit random walk in your question, or do you want a range of [-8, 7], which is what you get when you treat a 4-bit chunk of bits as a two's complement integer?
If the former, the random walk won't generate a uniform distribution, and you will need to tackle the problem in a different way.
In this case, to generate a uniform random number in the range [-4, 4], do the following:
Take 4 bits of the random bit generator and treat them as an integer in [0, 15);
If the integer is greater than 8, go to step 1.
Subtract 4 from the integer and output it.
This algorithm uses rejection sampling, but is variable-time (thus is not appropriate whenever timing differences can be exploited in a security attack). Numbers in other ranges are similarly generated, but the details are too involved to describe in this answer. See my article on random number generation methods for details.
Based on the code you've shown me, your approach to building up bytes, ints, and longs is highly error-prone. For example, a better way to build up an 8-bit byte to achieve what you want is as follows (keeping in mind that I am not very familiar with Kotlin, so the syntax may be wrong):
val i = 0
val b = 0
for (i = 0; i < 8; i++) {
b = b << 1; // Shift old bits
if (bitStringBuilder[i] == '1') {
b = b | 1; // Set new bit
} else {
b = b | 0; // Don't set new bit
}
}
value = (b as byte) as T
Also, if MediatorLiveData is not thread safe, then neither is your approach to gathering bits using a StringBuilder (especially because StringBuilder is not thread safe).
The approach you suggest, combining eight bits of the boolean generator to make one uniform integer, will work in theory. However, in practice there are several issues:
You don't mention what kind of hardware it is. In most cases, the hardware won't be likely to generate uniformly random Boolean bits unless the hardware is a so-called true random number generator designed for this purpose. For example, the hardware might generate uniformly distributed bits but have periodic behavior.
Entropy means how hard it is to predict the values a generator produces, compared to ideal random values. For example, a 64-bit data block with 32 bits of entropy is as hard to predict as an ideal random 32-bit data block. Characterizing a hardware device's entropy (or ability to produce unpredictable values) is far from trivial. Among other things, this involves entropy tests that have to be done across the full range of operating conditions suitable for the hardware (e.g., temperature, voltage).
Most hardware cannot produce uniform random values, so usually an additional step, called randomness extraction, entropy extraction, unbiasing, whitening, or deskewing, is done to transform the values the hardware generates into uniformly distributed random numbers. However, it works best if the hardware's entropy is characterized first (see previous point).
Finally, you still have to test whether the whole process delivers numbers that are "adequately random" for your purposes. There are several statistical tests that attempt to do so, such as NIST's Statistical Test Suite or TestU01.
For more information, see "Nondeterministic Sources and Seed Generation".
After your edits to this page, it seems you're going about the problem the wrong way. To produce a uniform random number, you don't add uniformly distributed random bits (e.g., bit() + bit() + bit()), but concatenate them (e.g., (bit() << 2) | (bit() << 1) | bit()). However, again, this will work in theory, but not in practice, for the reasons I mention above.
I have a multidimensional array (3D matrix) of unknown size, where each element in this matrix is of type short int.
The size of the matrix can be approximated to be around 10 x 10 x 1,000,000.
As I see it I have two options: Mutable Array (Objective-c) or Variable Array (c).
Are there any difference in reading writing to these arrays?
How large will these files become when I save to file?
Any advice would be gratefully accepted.
Provided you know the size of the array at the point of creation, i.e. you don't need to dynamically change the bounds, then a C array of short int with these dimensions will win easily - for reasons such as no encoding of values as objects and direct indexing.
If you write the array in binary to a file then it will just be the number of elements multiplied by sizeof(short int) without any overhead. If you need to also stored the dimensions that is 3 * sizeof(int) - 12 or 24 bytes.
The mutable array will be slower (albeit not by much) since its built on a C array. How large will the file be when you save this array?
It will take you more than 10x10x10000000 bytes because you'll have to encode it in a way where you can recall the matrix. This part is really up to you. For a 3D array, you'll have to use a special character/format in order to denote a 3D array. It depends on how you want to do this, but it will take 1 byte for every digit of every number + 1 char for the space you'll put between elements in the same row + (1 NL For every 2nd dimension in your array * n) + (1 other character for 3d values * n *n)
It might be easier to Stick each Row into its own file, and then stick the columns below it like you normally would. Then in a new file, I would start putting the 3d elements such that each line lines up with the column number of the 2nd dimension. That's just me though, its up to you.
i'm working on a simple multiplayer game that receives a random 4x4 matrix from a server and extracting a shape from it
for example:
XXOO
XXOX
XOOX
XXXX
OXOO
XXOO
XOOO
OXXX
so in the first matrix the shape i want to parse is this:
oo
o
oo
and the 2nd:
oo
oo
ooo
i know there must be an algorithm for this because i saw this kind of behavior on some puzzle games but i have no idea how to go about to detect them or even have an idea where to start
so my question is:how do i detect what shape is in the matrix and how do i differentiate between multiple colors? (aka..it doesn't come only in x and o..it comes in a maximum of 4)
note: the shape must be a minimum of 4 blocks
It's not fully clear what you want to do, but it seems like you want to extract some sort of shape (based on the second example, top left O not being included). I'd think about traversing the array, and for each cell, check for neighboring Os. Discount the duplicates you're likely to count (maybe by only looking at neighbors below and to the right of the cell you're examining). Then, if it meets whatever criteria you want for a 'shape'. Maybe if you give more examples or a better description, we can be more accurate. Or give it a shot yourself and post where you get stuck.
May be you can try this :
1. Assign integer values to each position in the matrix, like this, [1,2,3,4
5,6,7,8
9,10,11,12
13,14,15,16].
2.Read the position of zeroes. (I guess, shapes correpond to '0' s located either horizontally or vertically aligned). Store them in an array. So, for first case your array will read [3,4,7,11,10]
2. Then Start 'drawing' the shape.
1. First value 3. so shape= 0.
2. Next value 4. Check if it is consecutive to any other value in the array. that value is 3. so the shape = 00
3. Next val= 7. Is it consecutive ? no. Is it 4 more than any other value? yes, 3. So it goes below 3. shape= 00
0
4. Next 11, similar to step3, it goes below 7. shape= 00
0
0
5. Next 10, it is one less than 11, so it is placed before 11. shape= 00
0
00.
You can do some optimizations, like for consecutive, only check the prev. val in the array.
For vertical, check only four prev values.
Also, dont forget to special conditions for boundary values like 4 & 5. I dont they will make a shape.
This question comes from a homework assignment I was given. You can base your storage system off of one of the three following formats:
DD MM SS.S
DD MM.MMM
DD.DDDDD
You want to maximize the amount of data you can store by using as few bytes as possible.
My solution is based off the first format. I used 3 bytes for latitude: 8 bits for the DD (-90 to 90), 6 bits for the MM (0-59), and 10 bits for the SS.S (0-59.9). I then used 25 bits for the longitude: 9 bits for the DDD (-180 to 180), 6 bits for the MM, and 10 for the SS.S. This solution doesn't fit nicely on a byte border, but I figured the next reading can be stored immediately following the previous one, and 8 readings would use only 49 bytes.
I'm curious what methods others can come up. Is there a more efficient method to storing this data? As a note, I considered an offset based storage, but the problem gave no indication of how much the values may change between readings, so I'm assuming any change is possible.
Your suggested method is not optimal. You are using 10 bits (1024 possible values) to store a value in the range (0..599). This is a waste of space.
If you'll use 3 bytes for latitude, you should map the range [0, 2^24-1] to the range [-90, 90]. Hence each of the 2^24 values represents 180/2^24 degrees, which is 0.086 seconds.
If you want only 0.1 second accuracy, you'll need 23 bits for latitudes and 24 bits for longitudes (you'll get 0.077 seconds accuracy). That's 47 bit total instead of your 49 bits, with better accuracy.
Can we do even better?
The exact number of bits needed for 0.1 second accuracy is log2(180*60*60*10 * 360*60*60*10) < 46.256. Which means that you can use 46256 bits (5782 bytes) to store 1000 (lat,lon) pairs, but the mathematics involved will require dealing with very large integers.
Can we do even better?
It depends. If your data set has concentrations, you can store only some points and relative distances from these points, using less bits. Clustering algorithms should be used.
Sticking to existing technology:
If you used half precision floating point numbers to store only the DD.DDDDD data, you can be a lot more space-efficent, but you'd have to accept an exponent bias of 15, which means: The coordinates stored might not be exact, but at an offset from the original value.
This is due to the way floating point numbers are stored, essentially: A normalized significant is multiplied by an exponent to result in a number, instead of just storing a single value (as in integer numbers, the way you calculated the numbers for your solution).
The next highest commonly used floating point number mechanism uses 32 bits (the type "float" in many programming languages) - still efficient, but larger than your custom format.
If, however, you would design your own custom floating point type as well, and you gradually added more bits, your results would become more exact and it would STILL be more efficient than the solution you first found. Just play around with the number of bits used for significant and exponent, and find out how close your fp approximations come to the desired result in degrees!
Well, if this is for a large number of readings, then you may try a differential approach. Start with an absolute location, and then start saving incremental changes, which should ideally require less bits, depending on the nature of the changes. This is effectively compressing the stream. But somehow I don't think that's what this homework is about.
Trying to understand an fft (Fast Fourier Transform) routine I'm using (stealing)(recycling)
Input is an array of 512 data points which are a sample waveform.
Test data is generated into this array. fft transforms this array into frequency domain.
Trying to understand relationship between freq, period, sample rate and position in fft array. I'll illustrate with examples:
========================================
Sample rate is 1000 samples/s.
Generate a set of samples at 10Hz.
Input array has peak values at arr(28), arr(128), arr(228) ...
period = 100 sample points
peak value in fft array is at index 6 (excluding a huge value at 0)
========================================
Sample rate is 8000 samples/s
Generate set of samples at 440Hz
Input array peak values include arr(7), arr(25), arr(43), arr(61) ...
period = 18 sample points
peak value in fft array is at index 29 (excluding a huge value at 0)
========================================
How do I relate the index of the peak in the fft array to frequency ?
If you ignore the imaginary part, the frequency distribution is linear across bins:
Frequency#i = (Sampling rate/2)*(i/Nbins).
So for your first example, assumming you had 256 bins, the largest bin corresponds to a frequency of 1000/2 * 6/256 = 11.7 Hz.
Since your input was 10Hz, I'd guess that bin 5 (9.7Hz) also had a big component.
To get better accuracy, you need to take more samples, to get smaller bins.
Your second example gives 8000/2*29/256 = 453Hz. Again, close, but you need more bins.
Your resolution here is only 4000/256 = 15.6Hz.
It would be helpful if you were to provide your sample dataset.
My guess would be that you have what are called sampling artifacts. The strong signal at DC ( frequency 0 ) suggests that this is the case.
You should always ensure that the average value in your input data is zero - find the average and subtract it from each sample point before invoking the fft is good practice.
Along the same lines, you have to be careful about the sampling window artifact. It is important that the first and last data point are close to zero because otherwise the "step" from outside to inside the sampling window has the effect of injecting a whole lot of energy at different frequencies.
The bottom line is that doing an fft analysis requires more care than simply recycling a fft routine found somewhere.
Here are the first 100 sample points of a 10Hz signal as described in the question, massaged to avoid sampling artifacts
> sinx[1:100]
[1] 0.000000e+00 6.279052e-02 1.253332e-01 1.873813e-01 2.486899e-01 3.090170e-01 3.681246e-01 4.257793e-01 4.817537e-01 5.358268e-01
[11] 5.877853e-01 6.374240e-01 6.845471e-01 7.289686e-01 7.705132e-01 8.090170e-01 8.443279e-01 8.763067e-01 9.048271e-01 9.297765e-01
[21] 9.510565e-01 9.685832e-01 9.822873e-01 9.921147e-01 9.980267e-01 1.000000e+00 9.980267e-01 9.921147e-01 9.822873e-01 9.685832e-01
[31] 9.510565e-01 9.297765e-01 9.048271e-01 8.763067e-01 8.443279e-01 8.090170e-01 7.705132e-01 7.289686e-01 6.845471e-01 6.374240e-01
[41] 5.877853e-01 5.358268e-01 4.817537e-01 4.257793e-01 3.681246e-01 3.090170e-01 2.486899e-01 1.873813e-01 1.253332e-01 6.279052e-02
[51] -2.542075e-15 -6.279052e-02 -1.253332e-01 -1.873813e-01 -2.486899e-01 -3.090170e-01 -3.681246e-01 -4.257793e-01 -4.817537e-01 -5.358268e-01
[61] -5.877853e-01 -6.374240e-01 -6.845471e-01 -7.289686e-01 -7.705132e-01 -8.090170e-01 -8.443279e-01 -8.763067e-01 -9.048271e-01 -9.297765e-01
[71] -9.510565e-01 -9.685832e-01 -9.822873e-01 -9.921147e-01 -9.980267e-01 -1.000000e+00 -9.980267e-01 -9.921147e-01 -9.822873e-01 -9.685832e-01
[81] -9.510565e-01 -9.297765e-01 -9.048271e-01 -8.763067e-01 -8.443279e-01 -8.090170e-01 -7.705132e-01 -7.289686e-01 -6.845471e-01 -6.374240e-01
[91] -5.877853e-01 -5.358268e-01 -4.817537e-01 -4.257793e-01 -3.681246e-01 -3.090170e-01 -2.486899e-01 -1.873813e-01 -1.253332e-01 -6.279052e-02
And here is the resulting absolute values of the fft frequency domain
[1] 7.160038e-13 1.008741e-01 2.080408e-01 3.291725e-01 4.753899e-01 6.653660e-01 9.352601e-01 1.368212e+00 2.211653e+00 4.691243e+00 5.001674e+02
[12] 5.293086e+00 2.742218e+00 1.891330e+00 1.462830e+00 1.203175e+00 1.028079e+00 9.014559e-01 8.052577e-01 7.294489e-01
I'm a little rusty too on math and signal processing but with the additional info I can give it a shot.
If you want to know the signal energy per bin you need the magnitude of the complex output. So just looking at the real output is not enough. Even when the input is only real numbers. For every bin the magnitude of the output is sqrt(real^2 + imag^2), just like pythagoras :-)
bins 0 to 449 are positive frequencies from 0 Hz to 500 Hz. bins 500 to 1000 are negative frequencies and should be the same as the positive for a real signal. If you process one buffer every second frequencies and array indices line up nicely. So the peak at index 6 corresponds with 6Hz so that's a bit strange. This might be because you're only looking at the real output data and the real and imaginary data combine to give an expected peak at index 10. The frequencies should map linearly to the bins.
The peaks at 0 indicates a DC offset.
It's been some time since I've done FFT's but here's what I remember
FFT usually takes complex numbers as input and output. So I'm not really sure how the real and imaginary part of the input and output map to the arrays.
I don't really understand what you're doing. In the first example you say you process sample buffers at 10Hz for a sample rate of 1000 Hz? So you should have 10 buffers per second with 100 samples each. I don't get how your input array can be at least 228 samples long.
Usually the first half of the output buffer are frequency bins from 0 frequency (=dc offset) to 1/2 sample rate. and the 2nd half are negative frequencies. if your input is only real data with 0 for the imaginary signal positive and negative frequencies are the same. The relationship of real/imaginary signal on the output contains phase information from your input signal.
The frequency for bin i is i * (samplerate / n), where n is the number of samples in the FFT's input window.
If you're handling audio, since pitch is proportional to log of frequency, the pitch resolution of the bins increases as the frequency does -- it's hard to resolve low frequency signals accurately. To do so you need to use larger FFT windows, which reduces time resolution. There is a tradeoff of frequency against time resolution for a given sample rate.
You mention a bin with a large value at 0 -- this is the bin with frequency 0, i.e. the DC component. If this is large, then presumably your values are generally positive. Bin n/2 (in your case 256) is the Nyquist frequency, half the sample rate, which is the highest frequency that can be resolved in the sampled signal at this rate.
If the signal is real, then bins n/2+1 to n-1 will contain the complex conjugates of bins n/2-1 to 1, respectively. The DC value only appears once.
The samples are, as others have said, equally spaced in the frequency domain (not logarithmic).
For example 1, you should get this:
alt text http://home.comcast.net/~kootsoop/images/SINE1.jpg
For the other example you should get
alt text http://home.comcast.net/~kootsoop/images/SINE2.jpg
So your answers both appear to be correct regarding the peak location.
What I'm not getting is the large DC component. Are you sure you are generating a sine wave as the input? Does the input go negative? For a sinewave, the DC should be close to zero provided you get enough cycles.
Another avenue is to craft a Goertzel's Algorithm of each note center frequency you are looking for.
Once you get one implementation of the algorithm working you can make it such that it takes parameters to set it's center frequency. With that you could easily run 88 of them or what ever you need in a collection and scan for the peak value.
The Goertzel Algorithm is basically a single bin FFT. Using this method you can place your bins logarithmically as musical notes naturally go.
Some pseudo code from Wikipedia:
s_prev = 0
s_prev2 = 0
coeff = 2*cos(2*PI*normalized_frequency);
for each sample, x[n],
s = x[n] + coeff*s_prev - s_prev2;
s_prev2 = s_prev;
s_prev = s;
end
power = s_prev2*s_prev2 + s_prev*s_prev - coeff*s_prev2*s_prev;
The two variables representing the previous two samples are maintained for the next iteration. This can be then used in a streaming application. I thinks perhaps the power calculation should be inside the loop as well. (However it is not depicted as such in the Wiki article.)
In the tone detection case there would be 88 different coeficients, 88 pairs of previous samples and would result in 88 power output samples indicating the relative level in that frequency bin.
WaveyDavey says that he's capturing sound from a mic, thru the audio hardware of his computer, BUT that his results are not zero-centered. This sounds like a problem with the hardware. It SHOULD BE zero-centered.
When the room is quiet, the stream of values coming from the sound API should be very close to 0 amplitude, with slight +- variations for ambient noise. If a vibratory sound is present in the room (e.g. a piano, a flute, a voice) the data stream should show a fundamentally sinusoidal-based wave that goes both positive and negative, and averages near zero. If this is not the case, the system has some funk going on!
-Rick