Iam working on a real time data acquisition. In that I should send a string of ASCII characters in order to initiate the communication between the device and the laptop. This is the scenario.
Coming to the point, I want to know how to manually calculate the CRC for the following sequence 16,2,71,16,3,using CCITT polynomial i.e 0x1021.
I used online calculator for calculating the CRC using the following link.
http://www.lammertbies.nl/comm/info/crc-calculation.html
And now since that I know the values obtained in the online calculations and I am planning to manually verify them.Please help me with the issue if anyone is aware of the method used to calculate CRC.
Any help is appreciated.
Tnaks in advance.
Related
Final:Result after calculating and displaying the differenceI am new to VVC and I am going through the reference software's code trying to understand it. I have encoded and decoded videos using the reference software. I want to extract the bitstream from it, I want to know the number of bits there are in each macroblock. I am not sure which class I should be working with, for now I am looking at, mv.cpp, QuantRDOQ.cpp, and TrQuant.cpp.
I am afraid to mess the code up completely, I don't know where to add what lines of code. Start: Result after calculating and displaying the difference
P.S. The linked pictures are after my problem has been solved, I attached these pictures because of my query in the comments.
As the error says, getNumBins() is not supported by the CABAC estimator. So you should make sure you call it "only" during the encoding, and not during the RDO.
This should do the job:
if (isEncoding())
before = m_BinEncoder.getNumBins()
coding_unit( cu, partitioner, cuCtx );
if (isEncoding())
{
after = m_BinEncoder.getNumBins();
diff = after - before;
}
The simpleset solution that I'm aware of is at the encoder side.
The trick is to compute the difference in the number of written bits "before" and "after" encoding a Coding Unit (CU) (aka macroblock). This stuff happens in the CABACWriter.cpp file.
You should go to to coding_tree() function, where coding_unit() function is called, which is responsible for context-coding all syntax elementes in the current CU.
There, you may call the function getNumBins() twice: once before and once after coding_unit(). The difference of the two value should do the job for you.
I am an amateur wanted to take part in competition.Our first task was to read an image from a sd card, which we have achieved and we have stored in the form of binary array. Now we have to guide robot using this array to follow the map(image) in the maze.
Our team search on it but we are unable to find a solution. it is said that we can apply any version of Dijkstra's algorithm but for that we have to built the connectivity map.If you know any other way , please let us know. Since we are 1st year engineering students we don't know how to implement these algorithm. Can any one guide s to a achieve this task. we shall be very thankful.
Here is the picture of the array which we have read from bmp(image) file. Our robot has to enter from top and stop at "x" pattern made by array then exit. How can we use the binary data of 1's and 0's to mobilize the robot? We are using Arduino as our micro controller.
Link to original picture
This is the question:
Determine if the Hamming codes (15,10), (14,10) and (13,10) can correct a single error (SEC), detect a single error (SED) or detect double bit errors (DED).
I do know how Hamming distance work and how you can detect an error if you have the data-word that you want to transmit. But I don't know how to do it without the data-word.
Only for SEC which has the formula:
2^m > m+k+1
where
m = check bits
k = data bits
But is the any formulas for SED and DED? I have searched google all day long without any success.
I learned how to solve it checks the Hamming codes through this Youtube clip. there is no effective way. wish there was a faster way to solve the problem.
https://www.youtube.com/watch?v=JAMLuxdHH8o
This is not responding properly to the question because I know what the user
"user3314356" needs, but I could not comment because I did not have 50 reputation points!.
First of all, sorry for my poor English. I am writting video streaming server in C++. I have multiple mpeg2-ts files (movies and advertisements) which I need to stream via HTTP as one single TS-FILE. The problem is that every mpeg-ts file has its own timestamps (PCR, PTS, DTS). And, as I understand, to make a continuous streaming flow, every new PCR (PTS, DTS) value should continue from the last PCR (PTS, DTS) value.
Here is a picture for better understanding of what I am saying about: http://i.stack.imgur.com/vL1m6.png (I can't include my picture directly in the message. Sorry)
I need to replace pcr`1, pcr`2, pcr`3 timestamps with the new ones. For example, I sent ts-packet containing the pcr3 timestamp and after a few more ts packets (not containing any value of PCR) I want to insert my advertisement. And my question is: how do I calculate the new values for pcr`1, pcr`2, pcr`3 and so on?
Is it correct to calculate the bitrate of the current video and then divide the amount of bits that the program have sent since the last PCR timestamp (in our case, it's pcr3) by this bitrate? I mean the following: (new timestamp) = (previous timestamp) + (the amount of bits) / (bitrate). Or is there a more efficient way to do it?
As for PTS and DTS timestamps, I read here that these timestamps can be non-linear. Is it will be correct to calculate it relative to the last original PCR that I received? I mean:
pts_new = (original_pts - last_original_pcr) + pcr_new.
dts_new = (original_dts - last_original_pcr) + pcr_new.
(original_pts - last_original_pcr) is the difference between pts and pcr values
pcr_new is the last modified pcr value
My program can read and edit these timestamps in mpeg-ts stream. Fortunately, there's a lot of literature on how to do it. But how do I calculate the new values for these timestamps?
I have just started to learn the specification of mpeg2-ts, and please correct my mistakes if I am wrong in something.
Thanks in advance. Hope you understood me.
Mpeg2 "Splicing" is an art-form, and is much more complicated than concatenating two streams. It requires manipulations which many companies have patented (http://www.google.com/patents/US6380991, http://www.google.com/patents/US6806909, http://www.google.com/patents/US6993081)
to answer some of your questions:
your calculation of the next pcr looks ok, although you need take into account many compliancy issues (etr290 for example)
for DTS/PTS you have much more work to do. the most basic splice will just restamp the ad's pts/dts in such a way that they continue from the last timestamp of the first TS.
ad first timestamp = last timestamp + frame interval
the trick lies within making sure that you have no "holes" in either the presentation time stamps or the decoding timestamps. This is the difficult part and requires deep understanding in MPEG2 buffers (tstd, eb, mb).
Good luck.
Sorry to ask a similar question to the one i asked before (FFT Problem (Returns random results)), but i've looked up pitch detection and autocorrelation and have found some code for pitch detection using autocorrelation.
Im trying to do pitch detection of a users singing. Problem is, it keeps returning random results. I've got some code from http://code.google.com/p/yaalp/ which i've converted to C++ and modified (below). My sample rate is 2048, and data size is 1024. I'm detecting pitch of both a sine wave and mic input. The frequency of the sine wave is 726.0, and its detecting it to be 722.950820 (which im ok with), but its detecting the pitch of the mic as a random number from around 100 to around 1050.
I'm now using a High pass filter to remove the DC offset, but it's not working. Am i doing it right, and if so, what else can i do to fix it? Any help would be greatly appreciated!
(Fixed)
Thanks,
Niall.
Edit: Changed the code to implement a high pass filter with a cutoff of 30hz (from What Are High-Pass and Low-Pass Filters?, can anyone tell me how to convert the low-pass filter using convolution to a high-pass one?) but it's still returning random results. Plugging it into a VST host and using VST plugins to compare spectrums isn't an option to me unfortunately.
Edit: Fixed, thanks for everyones help, but I never got it to work, now using new code.
I am no sound expert, but if you are sampling with 44100 (I guess samples per second) and use 1024 datapoints. You are working with about 1/40th of a second worth of data. I doesn't surprise me that the current pitch varies a lot, depending on which piece you pick. If you want to find the average or main pitch of a voice, I'd expect to need about 1second worth of data.
At 44.1 kHz sampling frequency, 1024 samples is only a little bit over 23 ms worth of data. Isn't it possible that this is simply insufficient data in order to compute the pitch of a human singer?
I mean, the sound I can make that lasts for 23 ms is probably not something I have a lot of pitch-control over; I would expect this kind of measurement to be done over slighly longer periods of time.
The problem is in your findBestCandidates() function:
Inside this function you access the 'inputs' array from 0 up to 'length - 1'.
When you call this function inside detectPitchCalculation() function 'inputs' is 'results' and 'length' is 'nHiPeriodInSamples'.
But 'results' is only allocated and filled up to 'nHiPeriodInSamples - nLowPeriodInSamples - 1'.
So if 'nLowPeriodInSamples' is greater 0 you access unallocated and random memory inside the findBestCandidates() function!
EDIT:
Another bug is that you fill each 'nResolution' entry of the 'results' array in detectPitchCalculation() function but access each entry in the findBestCandidates() function (via the 'inputs' argument). But since you call detectPitchCalculation() with a 'nResolution=1' this does not explain your specific problem...so I will look a little bit more. But it would definitely a problem if you call it with higher resolutions.
I don't see the problem in you code, but I'm no good in C. But I'd try the following to find the problem:
run with data where the result in known, e.g. with sin(x) as input
run it with small data size (e.g. 2)
Compare the results with known correct ones. You should be able to find those on the internet, or do them by hand.
If random means: same input, different output, you most probably have some bug in the initialisation of variables. Use a debugger and known input to check, that all variables, especially all elements of arrays are properly initialized.