I am using HoughLinesP function in OpenCV. After reading the documentation here, I am confused regarding the necessity of two parameters "threshold" and "minLineLength".
Documentation says:
threshold – Accumulator threshold parameter. Only those lines are
returned that get enough votes ( >threshold ).
minLineLength – Minimum line length. Line segments shorter than that are rejected.
Isn't one of them redundant? Isn't the number of votes a line gets is equal to number of pixels it contains? Given that "minLineLength" specifies which lines to reject, what is the use of "threshold"? Am I missing something here. A clarification would be appreciated.
If the cells of accumulator are wide and there are many closely spaced short lines, then the count of accumulator would be high. The min_line_length would help solve such issues.
Hope this clarifies the question at hand.
Okay i read the documentation and how Randomized Hough Transform (will call it RHT) works and i have this idea, it might not be right though.
In RHT not all the binary points are taken into consideration, right. That's why it is faster. So, threshold means in the accumulator how many votes needed to consider this (rho, theta) as a line, this is the easy one.
The second one is a little bit confusing, but here is my though, minLineLength might be the length of the line calculated from the points that voted for it. Meaning, if we have a 5 points to vote for a certain line and all these points are so close, then the lineLength will be short may be 5 pixels. On the other hand, if 3 points voted for another line and the points are far a part, then the lineLength is large may be 20 pixels.
So we can't consider the voting threshold to be equal to the lineLength because not all binary points are taken into the calculation.
Related
I'm new to Z3 and as an exercise attempted a magic square solver by adapting an existing sudoku solver (http://lauri.võsandi.com/tub/qaoes/z3.html). I supply no facts (i.e. no specific numbers in specific boxes) other than the sum of all rows, columns and main diagonals are equal and entries are distinct and in the range [1,N*N]. It works fine for squares up to size 4. Any higher though and I give up waiting for a solution.
Is this normal? Or would experienced z3 programmers suggest my implementation has issues as problems of this size should be solvable?
Thanks.
You might have better results by expressing the entry in each cell as a BitVec() variable rather than an Int() variable.
See https://github.com/0vercl0k/z3-playground/blob/master/magic_square_z3.py for an example implementation. That implementation is able to find a solution for a 5x5 magic square in 1 second, a 6x6 magic square in 13 seconds, and a 7x7 magic square in 24 seconds (on my computer), so it seems to be doing significantly better than your formulation.
I'm doing a DIP project. I want to count the total number of words in each paper using Image Processing.
The original image is:
I did some pre-processing and produced the image below:
My idea to count the total number of words in each paper is to detect the digits inside blobs.
So please guide me. how can I count the words in this image? What's your idea?
Thanks.
Using the Digits inside blobs/circles is a good problem definition. I would recommend doing a circle hough transform and only looking for circles of a certain radius and then count the number of circles detected. You'll have to figure out what your radius is in pixels but this might be a good starting point. Good luck
If all pages are somewhat cleanly separated with one definition per line, you could take a very simple approach of counting the filled lines. First detect the list on the page to ignore irrelevant markings (green box) - does not have to exactly detect the edge so long as the bounds are no bigger than the list.
Then look for horizontal lines of pixels with no marking on them, or no dark value greater than X darkness. This is illustrated below with the pink horizontal lines. Lastly count the filled lines (any discrete section of horizontal lines which is not empty) and you have your number of definitions.
I used OpenCV's cv::findHomography API to calculate the homography matrix of two planar images.
The matched key points are extracted by SIFT and matched by BFMatcher. As I know, cv:findHomography use RANSAC iteration to find out the best four corresponding points to get the homography matrix.
So I draw the selected four pairs of points with the calculated contour using homograhy matrix of the edge of the object.
The result are as the links:
https://postimg.cc/image/5igwvfrx9/
As we can see, the selected matched points by RANSAC are correct, but the contour shows that the homography is not accurate.
But these test shows that, both the selected matched points and the homography are correct:
https://postimg.cc/image/dvjnvtm53/
My guess is that if the selected matched points are too close, the small error of the pixel position will lead to the significant error of the homography matrix. If the four points are in the corner of the image, then the shift of the matched points by 4-6 pixels still got good homography matrix.
(According the homogenous coordinate, I think it is reasonable, as the small error in the near plane will be amplified in the far away)
My question is:
1.Is my guess right?
2.Since the four matched points are generated by the RANSAC iteration, the overall error of all the keypoints are minimal. But How to get the stable homography, at least making the contour's mapping is correct? The theory proved that if the four corresponding points in a plane are found, the homography matrix should be calculated, but is there any trick in the engineer work?
I think you're right, and the proximity of the 4 points does not help the accuracy of the result. What you observe is maybe induced by numerical issues: the result may be locally correct for these 4 points but becomes worse when going further.
However, RANSAC will not help you here. The reason is simple: RANSAC is a robust estimation procedure that was designed to find the best point pairs among many correspondences (including some wrong ones). Then, in the inner loop of the RANSAC, a standard homography estimation is performed.
You can see RANSAC as a way to reject wrong point correspondences that would provoke a bad result.
Back to your problem:
What you really need is to have more points. In your examples, you use only 4 point correspondences, which is just enough to estimate an homography.
You will improve your result by providing more matches all over the target image. The problem then becomes over-determined, but a least squares solution can still be found by OpenCV. Furthermore, of there is some error either in the point correspondence process or in some point localization, RANSAC will be able to select the best ones and still give you a reliable result.
If RANSAC results in overfitting on some 4 points (as it seems to be the case in your example), try to relax the constraint by increasing the ransacReprojThreshold parameter.
Alternatively, you can either:
use a different estimator (the robust median CV_LMEDS is a good choice if there are few matching errors)
or use RANSAC in a first step with a large reprojection error (to get a rough estimate) in order to detect the spurious matchings then use LMEDS on the correct ones.
Just to extend #sansuiso's answer, with which I agree:
If you provide around 100 correspondences to RANSAC, probably you are getting more than 4 inliers from cvFindHomography. Check the status output parameter.
To obtain a good homography, you should have many more than 4 correspondences (note that 4 correspondences gives you an homography always), which are well distributed around the image and which are not linear. You can actually use a minimum number of inliers to decide whether the homography obtained is good enough.
Note that RANSAC finds a set of points that are consistent, but the way it has to say that that set is the best one (the reprojection error) is a bit limited. There is a RANSAC-like method, called MSAC, that uses a slightly different error measurement, check it out.
The bad news, in my experience, is that it is little likely to obtain a 100% precision homography most of the times. If you have several similar frames, it is possible that you see that homography changes a little between them.
There are tricks to improve this. For example, after obtaining a homography with RANSAC, you can use it to project your model into the image, and look for new correspondences, so you can find another homography that should be more accurate.
Your target has a lot of symmetric and similar elements. As other people mentioned (and you clarified later) the point spacing and point number can be a problem. Another problem is that SIFT is not designed to deal with significant perspective distortions that are present in your case. Try to track your object through smaller rotations and as was mentioned reproject it using the latest homography to make it look as close as possible to the original. This will also allow you to skip processing heavy SIFT and to use something as lightweight as FAST with cross correlation of image patches for matching.
You also may eventually come to understanding that using points is not enough. You have to use all that you got and this means lines or conics. If a homography transforms a point Pb = H* Pa it is easy to verify that in homogeneous coordinates line Lb = Henv.transposed * La. this directly follows from the equation La’.Pa = 0 = La’ * Hinv * H * Pa = La’ * Hinv * Pb = Lb’.Pb
The possible min. configurations is 1 line and three points or three lines and one point. Two lines and two points doesn’t work. You can use four lines or four points as well. Of course this means that you cannot use the openCV function anymore and has to write your own DLT and then non-linear optimization.
What are the applications of matrix addition to image processing?And also is there any application in image processing which modifies current pixel value based on values of neighbour pixels?
Strict addition is rare, but subtraction is more common, like when you subtract one image from its filtered one. For example you may subtract one image from its low pass filter and obtain its details.
Edit 1: Examples: image addition (averaging) for noise suppression. Subtraction for change enhancement.
Lots of image processing algorithms modify the pixels based on its neighbors, like many digital image filters.
Take for example the prewit filter which emphasizes horizontal edges. Its kernel is:
[1 1 1
0 0 0
-1 -1 -1]
Here the current pixel value will be replaced by the sum of its the north-west, north and north-east pixel value, minus the sum of its south-west, south and south-east pixel value.
Similar kernels, for other applications, include average, gaussian, laplacian, sobel, etc.. Its is also possible to compute fast rough distance maps.
Note 1: By the way, your acceptance rate is very low. People here may understand that you are ungrateful for the answers, and therefore won't bother to answer at all. Since you're new, I'm sure it's just because you need to understand how the system works. For any answer that has usefulness, click the arrow up. At some point, chose one of the answers as "the best one" and accept it (the green V tick sign.)
Note 2: TBH the question is actually quite vague, and would address a full book of digital image processing. In the future, try to be more specific (and, for stackoverflow, ask within programming).
All the best.
I want to put together a SDR system that tunes initially AM, later FM etc.
The system I am planning to use to do this will have a sine lookup table for Direct Digital Synthesis (DDS).
In order to tune properly I expect to need to be able to precisely control the frequency of the sine wave fed to the Mixer (multiplier in this case). I expect that linear interpolation will be close, but think a non-linear method will provide better results.
What is a good and fast interpolation method to use for sine tables. Multiplication and addition are cheap on the target system; division is costly.
Edit:
I am planning on implementing constants with multiply/shift functions to normalize the constants to scaled integers. Intermediate values will use wide adds, and multiplies will use 18 or 17 bits. Floating point "pre-computation" can be used, but not on the target platform. When I say "division is costly" I mean that it has to implemented using the multipliers and a lot of code. It's not unthinkable, but should be avoided. However, true floating point IEEE methods would take a significant amount of resources on this platform, as well as a custom implementation.
Any SDR experiences would be helpful.
If you don't get very good results with linear interpolation you can try the trigonometric relations.
Sum and Difference Formulas
sin(A+B)=sinA*cosB + cosA*sinB
sin(A-B)=sinA*cosB - cosA*sinB
cos(A+B)=cosA*cosB - sinA*sinB
cos(A-B)=cosA*cosB + sinA*sinB
and you can have precalculated sin and cos values for A, B ranges, ie
A range: 0, 10, 20, ... 90
B range: 0.01 ... 0.99
table interpolation for smooth functions = ick hurl bleah. IMHO I would only use table interpolation on some really weird function, or where you absolutely needed to ensure you avoid discontinuities (note that the derivatives for interpolated tables are discontinuous though). By the time you finish doing table lookups and the required interpolation code, you could have already evaluated a polynomial or two, at least if multiplication doesn't cause you too much heartburn.
IMHO you're much better off using Chebyshev approximation for each segment (e.g. -90 to +90 degrees, or -45 to +45 degrees, and then other segments of the same width) of the sine waveform, and picking the minimum degree polynomial that reduces your error to a desired value. If the segment is small enough you could get away with a quadratic or maybe even a linear polynomial; there's tradeoffs between accuracy, and # of segments, and degree of polynomial.
See my post in this other question, it'll save you the trouble of calculating coefficients (at least if you believe my math).
(edit: in case this wasn't clear, you do the Chebyshev approximation at design-time on your favorite high-powered PC, so that at run-time you can use a dirtbag microcontroller or FPGA or whatever with a simple polynomial of degree 1-4. Don't go over degree 4 unless you know what you're doing, 3 or below would be better.)
Why a table? This very fast function has its worst noise peak at -90db when the signal is at -20db. That's crazy good.
For resampling of audio, I always use one of the interpolators from the Elephant paper. This was discussed in a previous SO question.
If you're on a processor that doesn't have fp, you can still do these things, but they are harder. I've been there. I feel your pain. Good luck! I used to do conversions for fp to integer for fun, but now you'd have to pay me to do it. :-)
Cool online references that apply to your problem:
http://www.audiomulch.com/~rossb/code/sinusoids/
http://www.dattalo.com/technical/theory/sinewave.html
Edit: additional thoughts based on your comments
Since you're working on a tricky processor, maybe you should look into how to make your sine table have more angles to look up, but still keep it small.
Suppose you break a quadrant into 90 pieces (in reality, you'd probably use 256 pieces, but let's keep it 90 for familiarity and clarity). Encode those as 16 bits. That's 180 bytes of table so far.
Now, for every one of those degrees, we're going to have 9 (in reality probably 8 or 16) in-between points.
Let's take the range between 3 degrees and 4 degrees as an example.
sin(3)=0.052335956 //this will be in your table as a 16-bit number
sin(4)=0.069756474 //this will be in your table as a 16-bit number
so we're going to look at sin(3.1)
sin(3.1)=0.054978813 //we're going to be tricky and store the result
// in 8 bits as a percentage of the distance between
// sin(3) and sin(4)
What you want to do is figure out how sin(3.1) fits in between sin(3) and sin(4). If it's half way between, code that as a byte of 128. If it's a quarter of the way between, code that as 64.
That's an additional 90 bytes and you've encoded down to a tenth of a degree in 16-bit res in only 180+90*9 bytes. You can extend as needed (maybe going up to 32-bit angles and 16-bit tween angles) and linearly interpolate in between very quickly. To minimize storage space, you're taking advantage of the fact that consecutive values are close to each other.
Edit 2: better way to encode the in-between angles in a table
I just remembered that when I did this, I ended up very compactly expressing the difference between the expected value according to linear interpolation and the actual value. This error is always in the same direction.
I first calculated the maximum error in the range and then based the scale on that.
Worked great. I feel like I should do the code in a blog entry to illustrate. :-)
Interpolation in a sine table is effectively resampling. Obviously you can get perfect results by a single call to sin, so whatever your solution is it needs to outperform that. For fixed-filter resampling, you're still going to only have a fixed set of available points (a 3:1 upsampler means you'll have 2 new points available between each point in your table). How expensive is memory on the target system? My primary recommendation is simply improve the table resolution and use linear interpolation. You'll get the same results as a smaller table and simple upsample but with less computational overhead.
Have you considered using the Taylor series for the trig functions (found here)? This involves multiplication and division but depending on how your numbers are represented you may be able to turn the division into multiplication (or bit shifts if you're very lucky). You can compute as many terms of the series as you need and get your precision that way.
Alternately if this sine wave is going to be an analog signal at some point then you could just use a lookup table approach and use an analog filter to remove the sampling frequency from the resulting waveform. If your sampling frequency is 100 times the sine frequency it will be easy to remove. You'll need a variable filter to do this. I've never done such a thing but I know there's digital potentiometers that take a binary number and change their resistance. That could be the basis of a variable RC filter - probably with some op-amps for gain, etc.
Good luck!
People have written some amazingly clever code for quickly calculating sin() on systems with tiny amounts of memory that don't even have a hardware multiply instruction, much less a division instruction.
In order of increasing complexity:
Use a square wave. Many AM radios use square waves in their ring demodulator, and I fail to see why your AM demodulator requires anything more complicated.
Approximate sin() by looking up the "closest value" in a raw table of 256 values per quarter-cycle. Yes, you see horrible-looking stair-steps, but (with a little bit of analog filtering) this often works well. (In fact, this is often overkill, and a much shorter table is adequate).
Approximate sin() by looking up the 2 closest values in a raw table, and linearly interpolating between them.
Approximate sin() with 16 short, equally-spaced-in-x cubic splines per quarter-cycle "gives better than 16-bit precision" for sin(x).
Wikibooks: Fixed-Point Numbers links to some clever implementations of the last 3.