I got this code from google code :
void QBluetoothDeviceDiscoveryAgent::deviceDiscovered(const QBluetoothDeviceInfo &info)
QBluetoothDeviceInfo::rssi().
But how to get rssi distance from `QBluetoothServiceDiscoveryAgent ?
I tried with
QBluetoothServiceDiscoveryAgent serviceInfo;
quint i =serviceInfo.device().rssi();
here i = -43
how to convert it to distance?
I got the link
Understanding ibeacon distancing
but how to get the transmitter power? to calculate the distance according to formula?
Make sure you understood the implications of QBluetoothDeviceInfo::rssi(). Calling this functions returns immediately with the last stored value when the device was scanned last. If you only receive one advertisement-packet, which happens to be at e.x. -90dB, and then immediately connect, this function will keep returning -90 until you disconnect from it and scan it again. Connected devices usually don't send advertisement-packets so the RSSI you can read via Qt won't be updated during the connection.
As for proximity, it's not so easy to get good values. To accurately convert from RSSI to geometric distance you must know the sender's original/intended signal-strength (or TX-power-level == RSSI at 1m distance). This value will differ between devices. To make things worse, in practice it can also vary by a huge margin depending on things like the sender's battery-level, physical orientations of sender/receiver to eachother, quality of individual parts, random interference from other RF devices....
The BLE-folk has a blog explaining how you should do it. You can read it up here. The linked article doesn't read or assume the theoretical maximum RSSI of the sender but instead it propoposes to gather multiple RSSI-values over time (+ do some mean/mode filtering), and use the current mean-value in comparison with the previous value to determine if you are approaching or moving away from the sender. Paired with some fine-tuning using real-world data you gotta collect, plus documentation-reading and common-sense, you could probably develop a proximity calculation for many or even most sender-devices which would be accurate to about one meter or even less at close proximity. In the end it's a tradeoff between how many devices you wish to 'calibrate' for and those you are okay with having shifted values due to higher or lower TX-power-levels.
The downside being - you can't test for every possible device on the market and as I said earlier, different devices have different TX-power-levels. With this approach you can develop an algorithm to get pretty good measurements for devices which have approximately equal signal-configurations but others will seem far off. The article's author talks about creating different profiles for different vendors but that's not really gonna help (consider two identical beacons ("big/small"), one for large and one for small indoor locations - with RSSI alone you can't reliably determine if you're close to the small beacon or in medium range to the big one unless they identify themselves via GAP or otherwise (forget MAC-addresses if you plan to deploy on MacOS or iOS).
Also, prepare yourself for the joyride that is Android BLE development. Some vendors know that their BLE implementation is so terribly bad and broken, they even disabled the HCI-Logging-Feature on all their ROMs to hide it. Others can be BLE-nuked like Win98 by ethernet, back in the days.
Related
I am working user behavior project. Based on user interaction I have got some data. There is nice sequence which smoothly increases and decreases over the time. But there are little discrepancies, which are very bad. Please refer to graph below:
You can also find data here:
2.0789 2.09604 2.11472 2.13414 2.15609 2.17776 2.2021 2.22722 2.25019 2.27304 2.29724 2.31991 2.34285 2.36569 2.38682 2.40634 2.42068 2.43947 2.45099 2.46564 2.48385 2.49747 2.49031 2.51458 2.5149 2.52632 2.54689 2.56077 2.57821 2.57877 2.59104 2.57625 2.55987 2.5694 2.56244 2.56599 2.54696 2.52479 2.50345 2.48306 2.50934 2.4512 2.43586 2.40664 2.38721 2.3816 2.36415 2.33408 2.31225 2.28801 2.26583 2.24054 2.2135 2.19678 2.16366 2.13945 2.11102 2.08389 2.05533 2.02899 2.00373 1.9752 1.94862 1.91982 1.89125 1.86307 1.83539 1.80641 1.77946 1.75333 1.72765 1.70417 1.68106 1.65971 1.64032 1.62386 1.6034 1.5829 1.56022 1.54167 1.53141 1.52329 1.51128 1.52125 1.51127 1.50753 1.51494 1.51777 1.55563 1.56948 1.57866 1.60095 1.61939 1.64399 1.67643 1.70784 1.74259 1.7815 1.81939 1.84942 1.87731
1.89895 1.91676 1.92987
I would want to smooth out this sequence. The technique should be able to eliminate numbers with characteristic of X and Y, i.e. error in mono-increasing or mono-decreasing.
If not eliminate, technique should be able to shift them so that series is not affected by errors.
What I have tried and failed:
I tried to test difference between values. In some special cases it works, but for sequence as presented in this the distance between numbers is not such that I can cut out errors
I tried applying a counter, which is some X, then only change is accepted otherwise point is mapped to previous point only. Here I have great trouble deciding on value of X, because this is based on user-interaction, I am not really controller of it. If user interaction is such that its plot would be a zigzag pattern, I am ending up with 'no user movement data detected at all' situation.
Please share the techniques that you are aware of.
PS: Data made available in this example is a particular case. There is no typical pattern in which numbers are going to occure, but we expect some range to be continuous with all the examples. Solution I am seeking is generic.
I do not know how much effort you want to involve in this problem but if you want theoretical guaranties,
topological persistence seems well adapted to your problem imho.
Basically with that method, you can filtrate local maximum/minimum by fixing a scale
and there are theoritical proofs that says that if you sampling is
close from your function, then you extracts correct number of maximums with persistence.
You can see these slides (mainly pages 7-9 to get the idea) to get an idea of the method.
Basically, if you take your points as a landscape and imagine a watershed starting from maximum height and decreasing, you have some picks.
Every pick has a time where it is born which is the time where it becomes emerged and a time where it dies which is when it merges with an higher pick. Now a persistence diagram pictures a point for every pick where its x/y coordinates are its time of birth/death (by assumption the first pick does not die and is not shown).
If a pick is a global maximal, then it will be further from the diagonal in the persistence diagram than a local maximum pick. To remove local maximums you have to remove picks close to the diagonal. There are fours local maximums in your example as you can see with the persistence diagram of your data (thanks for providing the data btw) and two global ones (the first pick is not pictured in a persistence diagram):
If you noise your data like that :
You will still get a very decent persistence diagram that will allow you to filter local maximum as you want :
Please ask if you want more details or references.
Since you can not decide on a cut off frequency, and not even on the filter you want to use, I would implement several, and let the user set the parameters.
The first thing that I thought of is running average, and you can see that there are so many things to set, to get different outputs.
I have a neural network written in standard C++11 which I believe follows the back-propagation algorithm correctly (based on this). If I output the error in each step of the algorithm, however, it seems to oscillate without dampening over time. I've tried removing momentum entirely and choosing a very small learning rate (0.02), but it still oscillates at roughly the same amplitude per network (with each network having a different amplitude within a certain range).
Further, all inputs result in the same output (a problem I found posted here before, although for a different language. The author also mentions that he never got it working.)
The code can be found here.
To summarize how I have implemented the network:
Neurons hold the current weights to the neurons ahead of them, previous changes to those weights, and the sum of all inputs.
Neurons can have their value (sum of all inputs) accessed, or can output the result of passing said value through a given activation function.
NeuronLayers act as Neuron containers and set up the actual connections to the next layer.
NeuronLayers can send the actual outputs to the next layer (instead of pulling from the previous).
FFNeuralNetworks act as containers for NeuronLayers and manage forward-propagation, error calculation, and back-propagation. They can also simply process inputs.
The input layer of an FFNeuralNetwork sends its weighted values (value * weight) to the next layer. Each neuron in each layer afterwards outputs the weighted result of the activation function unless it is a bias, or the layer is the output layer (biases output the weighted value, the output layer simply passes the sum through the activation function).
Have I made a fundamental mistake in the implementation (a misunderstanding of the theory), or is there some simple bug I haven't found yet? If it would be a bug, where might it be?
Why might the error oscillate by the amount it does (around +-(0.2 +- learning rate)) even with a very low learning rate? Why might all the outputs be the same, no matter the input?
I've gone over most of it so much that I might be skipping over something, but I think I may have a plain misunderstanding of the theory.
It turns out I was just staring at the FFNeuralNetwork parts too much and accidentally used the wrong input set to confirm the correctness of the network. It actually does work correctly with the right learning rate, momentum, and number of iterations.
Specifically, in main, I was using inputs instead of a smaller array in to test the outputs of the network.
How I can get loudness level from raw data received from microphone in DirectShow?
IMediaSample keep data in bytes. And how I can read this bytes and get something?
Loudness is an aural quality, not a physic formula. There are many many definitions for it.
It's a also a temporal value. As a consequence, this value changes during the time.
The simplest implementation I remember I had seen some years ago, was simply putting a time out on the maximum value of the amplitude. But the log of the amplitude is surely better to approximate the ear sensitivity much closer.
You can also consider the power of the signal ( signal * signal ... but there are also more definitions that takes into account the frequency spectrum components...).
It's kitchen recipes. Choose the simplest.
Edit: it seems my answer was too fast and fuzzy, I probably mistake Volume and Loudness. this wikipedia article states there are units for measuring loudness. Sone and Phon.
You need to process data to calculate loudness out of raw bytes. One of the method is defined in BS.1770 : Algorithms to measure audio programme loudness and true-peak audio level specification and describes the algorithm involved.
This is to be done in C++ or C....
I know we can read the MP3s' meta data, but that information can be changed by anyone, can't it?
So is there a way to analyze a file's contents and compare it against another file and determine if it is in fact the same song?
edit
Lots of interesting things coming out that I hadn't thought of. Not at all a good idea to attempt this.
It's possible, but very hard.
Even the same original recording may well be encoded differently by different MP3 encoders or the same encoder with different settings... leading to different results when the MP3 is then decoded. You'd need to work out an aural model to "understand" how big the differences are, and make a judgement.
Then there's the matter of different recordings. If I sing "Once in Royal David's City" and Aled Jones sings it, are those the same song? What if there are two different versions of a song where one has slightly modified lyrics? The key could be different, it could be in a different vocal range - all kinds of things.
How different can two songs be but still count as "the same song"? Once you've decided that, then there's the small matter of implementing it ;)
If I really had to do this, my first attempt would be to take a Fourier transform of both songs and compare the histograms. You can use FFTW (http://www.fftw.org/) to take the Fourier transform, and then compare the histograms by summing the squares of the differences at each frequency. If the resultant sum is greater than some threshold (which you must determine by experimentation) then the songs are deemed to be different, otherwise they are the same.
No. Not SO simple.
You can check they contain the same encoded data, BUT:
Could be a different bitrate
Could be the same song, just a 1/100ths of a second off
In both cases the bytes would not match.
Basically, if a solution looks too simple to be true, it often is.
If you mean "same song" in the iTunes sense of "same recording", it would be possible to compares two audio files, but not by byte-by-byte comparison of an encoded file since even for the same format there are variables such as data rate and compression that are selected at time of encoding.
Also each encoding of the same recording may include different lead-in/lead-out timings, different amplitude and equalisation, and may have come from differing original sources (vinyl, CD, original master etc.). So you need a comparison method that takes all these variables into account, and even then you will end up with a 'likelihood' of a match rather than a definitive match.
If you genuinely mean "same song", i.e. any recording by any artist of the same composition and lyrics, then you are unlikely to get a high statistical correlation in most cases since pitch, tempo, range, instrumental arrangement will be very different.
In the "same recording" scenario, relatively simple signal processing and statistical techniques could be applied, in the "same song" scenario, AI techniques would need to be deployed, and even then the results I suspect would be poor.
If you want to compare MP3 files that originated from the same MP3, but have tagged with metadata differently, it would be straight forward to just compare the actual audio data. Since it originated from the same MP3 encoding, you should be able to do a byte by byte comparison. You would have to compare all byte. It should be sufficient to sample just a few to get a unique key that would be statistically almost impossible to find in another song.
If the files have been produced by different encoders, you would have to extract some "fuzzy" feature keys from the data and compare those keys. In a hurry I would probably construct an algorithm like this:
Decode audio to pulse-code modulation (wave) in a standard bit rate.
Find a fixed number of feature starting points using some dynamic location algorithm. For example find top 10 highest wave peaks ordered from beginning of wave or simply spread evenly across the wave (it would be a good idea to fix the first and last position dynamically though, since different encodings might not start and end at exactly the same point). An improvement would be to select feature points at positions in the wave that are not likely to be too repetitive.
Extract a set of one-dimensional feature key scalars from the feature points. For example, for each feature normalize the following n-sample values and count the number of zero-crossings, peak to average ratio, mean zero-crossing distance, signal-energy. The goal is to extract robust features that are relatively unique, while still characteristic even if some noise and distortion is added to the signal. This can obviously be improved almost infinitely.
Compare the extracted feature keys of the two files using some accuracy measurement (f.eks. 9 out of 10 feature extractions must match at least 99% on 4 out of 5 of their extracted feature keys).
The benefit of a feature extraction approach is that you can build a database of features for all your mp3-files and for a single file ask the question: What other media files have exactly or almost exactly the same feature as this one. The feature lookup could be implemented very efficiently with R*-trees or similar, which could be used to give you a fast distance measurement between the n-dimensional feature sets.
The above technique is essentially a variant of what is used in image search algorithms such as SIFT, which is probably the base of such application as Photosynth and Google Goggles. In image searching you filter the image for good candidate points for relatively unique features (such as corners of shapes), then you normalize the area around that feature to get normalized color, intensity, scale and direction of features. Finally you extract the features and search an n-dimensional database of features of other images and verify that found features in other images are geometrically positioned in the same pattern as in your search image. The technique for searching audio would be the same, only simpler, since audio is one dimensional.
Use the open source EchoPrint library to create a signature of the two audio files, and compare them with each other.
The library is very easy to use, and has clear examples on how to create the signatures.
http://echoprint.me/
You can even query their database with the signature and find matching song metadata (such as title, artist, etc).
I think the Fast Fourier-Transform (FFT) approach hinted by jstanley is pretty good for most use cases; in particular, it works for verifying that the two are the same release/ same recording by the same artist/ same bitrate / audio quality.
To be more explicit, sox and spek (via command line and GUI, respectively) can do this pretty painlessly.
Spek is pretty foolproof -- just open the software and point it to the two audio files in question.
sox can generate spectograms (FFTs) from the command line line so:
sox "$file" -n spectrogram -o "$outfile".
The result from either are two images; if they look basically identical, then for almost all intents and purposes, the two songs will be equivalent.
For example, I wanted to test if these two files:
Soundtrack to an imaginary film mixtape 2011.mp3
DJRUM - Sountrack to an imaginary film mixtape 2011 (for mary-anne hobbs).mp3
were the same. diff reported a difference in the binary files (perhaps due to metadata differences or minor encoding differences), but a quick glance at their spectrograms resolved it:
I'm going to write a program that plots data from a sensor connected to the computer. The sensor value is going to be plotted as a function of the time (sensor value on the y-axis, time on the x-axis). I want to be able to add new values to the plot in real time. What would be best to do this with in C++?
Edit: And by the way, the program will be running on a Linux machine
Are you particularly concerned about the C++ aspect? I've done 10Hz or so rate data without breaking a sweat by putting gnuplot into a read/plot/refresh loop or with LiveGraph with no issues.
Write a function that can plot a std::deque in a way you like, then .push_back() values from the sensor onto the queue as they come available, and .pop_front() values from the queue if it becomes too long for nice plotting.
The exact nature of your plotting function depends on your platform, needs, sense of esthetics, etc.
You can use ring buffers. In such buffer you have read position and write position. This way one thread can write to buffer and other read and plot a graph. For efficiency you usually end up writing your own framework.
Size of such buffer can be estimated using eg.: data delivery speed from sensor (40KHz?), size of one probe and time span you would like to keep for plotting purposes.
It also depends whether you would like to store such data uncompressed, store rendered plot - all for further offline analysis. In non-RTOS environment your "real-time" depends on processing speed: how fast you can retrieve/store/process and plot data. Usually it is near-real time efficiency.
You might want to check out RRDtool to see whether it meets your requirements.
RRDtool is a high performance data logging and graphing system for time series data.
I did a similar thing for a device that had a permeability sensor attached via RS232.
package bytes received from sensor into packets
use a collection (mainly a list) to store them
prevent the collection to go over a fixed size by trashing least recent values before new ones arrive
find a suitable graphics library to draw with (maybe SDL if you wanna keep it easy and cross-platform), but this choice depends on what kind of graph you need (ncurses may be enough)
last but not least: since you are using a sensor I suppose your approach will be multi-threaded so think about it and use a synchronized collection or a collection that allows adding values when other threads are retrieving them (so forgot iterators, maybe an array is enough)
Btw I think there are so many libraries, just search for them:
first
second
...
I assume that you will deploy this application on a RTOS. But, what will be the data rate and what are real-time requirements! Therefore, as written above, a simple solution may be more than enough. But, if you have hard-real time constraints everything changes drastically. A multi-threaded design with data pipes may solve your real-time problems.