What would be the best way to triangulate a wireless network passively. Are there tools available? Algorithms? Libraries?
My goal would be to create a relative map of various objects that sends or receive signals using signal strength (DB's), signal/noise ratio, signal phase, etc. from a few location points. With enough sampling, i'm guessing it would be possible to create a good 2d/3d map.
I'm searching for stuff in any language / platform.
Some keywords: wi-fi site survey, visualization, coverage, location, positioning
Thinking about using kismet to gather the data and then process it. Maybe Free Space Path Loss for RF in the 2.4Ghz range to calculate a relative distance. And optionally try to use RF obstacle attenuation estimation values (based on some user input) to give better estimates. Then use trilateration to generate possible relative coordinates.
You can't use the GPS technique because the timing is nothing like accurate enough.
The best you can do is Trilateration based on the signal strength from each base station and assume that range is proportional to signal.
You will probably need to force a connection to each base station in turn in order to measure the signal strength.
Interesting question. Initial thoughts were using output from something like the WiSpy spectrum analyzer. I like the idea of using a directional antenna. Looks like some research (may) be underway.
Instead of trilateration you could use bilinear interpolation. This is said to be better for non-linear distance vs. signal strength data like wifi in an urban environment would be. http://courses.cit.cornell.edu/ee476/FinalProjects/s2007/ayl26_ym82/ayl26_ym82/index.htm has the background math and the what I assume is AVR C for doing it with magnetic field sensors.
Using signal strength to judge distance could easily be thrown off by differences in materials blocking line-of-sight to each of the sampling points. It would probably be better to do the sampling with a directional antenna, and from each sampling point, find the bearing that maximizes signal strength to each device you want to locate. With this technique, you can use only two or three sampling locations, depending on the accuracy with which you can estimate the bearings.
Ars Technica has an article about this, citing the Fraunhofer Institute and Skyhooks Wireless. This technology is built into every iPhone and iPad.
Actually I think you should try using an algorithm like the GPS one (wikipedia).. of course you can simplify it according to your need, for example:
you need to install on every item that should broadcast its position (the navigation signal) an application that actually does it
you should use a different channel for every single item to be sure not to generate collisions (it depends also on how much you broadcast the signal)
so if you place at least 4 broadcasters you can triangulate on every client to allow it to calculate its position. Naturally the bcasters should be as much similar as possible in response..
by the way these are just ideas..
Related
Well, i want to ask if ADXL345 can be used to detect an Earthquake Occurrence based on its magnitude/intensity level. For more information, I want to used an accelerometer to create a Device that can detect the intensity/magnitude level of an Earthquake.
I have absolutely no experience in this field, but it looks useful and fascinating.
Questions are:
is this device able to detect medium scale earthquakes?
if yes, does anybody did it, available to share experiences?
if no to the previous, is there any guide which explains algorithms, calculations and mechanical plans?
That sensor is not suitable. It has 13 bit resolution at +-16g full range. That gives you a sensitivity of 0.002g for the lsb. In order to detect an earthquake directly below you, you need approx. a few milli-g (e.g. see here), even less for earthquakes with an epicentre elsewhere.
You want a sensor which is much more sensitive by a factor of 100 and probably with more resolution (better ADC), too.
(And you should have been able to do this quick google-search analysis yourself ;) )
Using accelerometers reading tells you nothing about the actual magnitude of the quake itself. It tells you the size of the quake at your location. Combining location and amplitude will give you a 'weighted' measurement, but that's still useless without a calibration curve. Without knowing what acceleration, at a certain distance, corresponds to what magnitude you will be unable to tell what the magnitude is. You can certainly conclude that your measured earthquake has a median amplitude of, say, 2000% of a non-earthquake reading, but you won't be able to turn it into a Richter measurement. To do that you'd need to take some data during earthquakes of known magnitude and then work out how acceleration, distance and magnitude are related for your device. You could alternatively use a scale like the Shindo (just Google it).
I wanted to know if there is any efficient way of finding the distance between 2 devices(a transmitter and a receiver) which is accurate to atleast the order of a couple of inches.
I am basically want to detect the movement of the transmitter from the receiver and how far it has moved from its original position.
I was thinking in terms of using a wireless hotspot/bluetooth connection. I cannot Use some form of audio/medium which can be detected by humans.
Could anybody help me with this?
To my mind, assuming there is no common synchronisation signal between the devices, there are 2 differents way to do this (not really easy):
1. Measure received power : some receivers provide RSSI (Received Signal Strength Indication). RSSI is a measure of how much power you received. If you know the transmitted power, you can estimate the transmission loss (from the transsmission channel) by taking different measure of RSSI at different distance. It will really depends on the channel (environment, frequency, throughput, ..), so don't change it for the measure. Once you got enough points, try to fit it by a curve. You can now predict distance by having RSSI.
2. Measure round trip time : this is called RADAR, and is really more difficult but is the classic way to measure distance and speed. Broadband systems (like WiFi) are better for this kind of measure. By the way you also can do the same with audio for short distances (SONAR), without being detected if you use frequencies higher than 20kHz.
I'm trying to measure the distance an object is pulled along a pulley track using arduino (or Rasp Pi) sensors. I have an object that is manually pulled, from a resting position, on a pulley system and I need to be able to track the distance it travels over one pull.
Example: The object traveled 90% (example) of the total pulley distance for 5/10 pulls.
Example: The object traveled 11.53 ft along the pulley.
See the image below for a visual diagram. I have two ideas, but I'm not an Arduino sensor expert. I'd love input on an elegant solution.
Use an arduino sensor to sense how many times the pulley rotates, and then use an equation to determine the distance.
Use a sensor that senses the distance pulled in cord, maybe each direction, and record that distance. I got this idea from pump sensors, no idea if a comparison exists.
Really just looking for advice on what sensors to use, how to implement them (general), and what type of metrics I could record.
There are at least two ways of doing this:
Use a rotary encoder, which will tell how many rotations the pulley did. There are two disadvantages: line can slip on the pulley, resulting in inaccurate measurement, and there is no way you can now at what actual position the object is. Rotary encoder will give you only position relative to the starting position (unless you only need one rotation of the pulley - then you can use absolute encoders)
Use a distance sensor (ultrasonic, or IR) to measure at what height is the object. That way you will know the exact position of the object, but you may have a problem if your object is too small, has a shape or surface that will impact the measurements, or if the space is limited and a sensor can pick up walls, or other surrounding objects. Also if the objects may swing on the rope, there will probably a problem.
Software implementation for all of these solutions should be pretty simple, just decide on type of sensors - there are plenty of tutorials for all of them.
Both task 1 and 2 can be done easily using a digital rotary encoder at very nominal cost. It can sense direction and distance of travel quite accurately.
I want to ask about I Beacon advertising, especially Tx Power.
I used two BLE module HM10 and HM11. I make one as a ibeacon (HM10). and other one used to connect and listen to HM10 broadcasting.
I used MCU ATmega32 AVR tied with HM11 and I used scanf function to read the broadcast. I want to extract the last byte (Tx Power). I want to measure the distance with AVR programming.
Could you tell me the algorithm?
The formula Apple uses to calculate a distance estimate to an iBeacon is not published. There are a number of alternative formulas including this one, based on a best fit power curve, that we wrote for the Android Beacon Library.
Further research we have done shows that the formula above basically works, but it has two main imperfections:
It does not work well for weaker beacon transmitters. With weaker broadcasts, the distance is underestimated.
It does not account for varying signal gains in receivers. Different receivers have different antennas and receivers which measure the same signals differently.
There is an ongoing discussion of the best formula here.
A bit late but hopefully useful to others. I have given up on Apple's "Accuracy" number; as #davidyoung points out, different devices will have different signal gains. Now I am not an engineer but more of a math and statistics person, so I have gone down the route of "fingerprinting" an indoor space instead. Essentially I read all RSSI from all beacons installed in a certain "venue". Some might not be within reach and therefore I just assume, in such cases, an RSSI of -95 dBm (which seems to be the floor past which a signal is not read any more). Such constituted array has the same beacons in the same positions at all times (even across app launches). I compute a 5 seconds moving average for each beacon (so a I se 5 arrays to do that). The resulting avg array is then shifted up by 95 units and normalised so that the sum of all of its values is one. If you want to tag an an indoor "point" you collect many of these normalised average arrays on that specific spot. I go ahead and construct a database of "spots". To forecast your proximity to any spot in a database you simply compute a quadratic distance of your current reading and the all of the fingerprints in the database.
Which beacons to use? At least class 2 in power. How many? At least a couple per room (put them in two adjacent corners, on the ceiling or high up).
The last step that you need to do is match the fingerprints with an x,y coordinate on your map. I never did this step, because I am mainly interested in proximity applications and not fully fingerprint and indoor space.
Perhaps the discussion above will serve you as a guidance on a technique that is used by many indoor location companies.
Disclosure: I have recently open sourced my code doing the above calculations.
In a previous question, I had asked Why can't I simply negate the source time domain amplitude values to produce a destructive noise signal?
One of the posters said that while simply producing a inverses polarity (negated) signal will work in theory, in practice it is not possible
So I am asking, what is the fundamental approach (in a sort of semi technical way) to active noise cancellation?
Secondly, why are most literature on this topic in frequency domain?
It's rather simple.
By the time you send your inverted signal, the noise has already been heard.
You need to look at what frequencies are being generated, and then produce the appropriate inverted signals of those to cancel them out.
Noise cancellation is prediction. Your algorithm has to predict what the sound of the noise will be at some time in the future (that time given by the system and audio time latencies), and then predict what signal will produce the opposite sound at that same point in the future (which your system will distort and delay, so you have to figure in the opposite distortion and delay).
You might be able to use several successive FFTs to determine which frequencies in the noise are not changing, and assume or calculate some probability that they will continue for a short time into the future.
If you know the frequency response curve of the speaker, you might be able to figure out the frequency amplitudes of a signal needed to match some predicted noise spectrum. The phase angle of a sinusoid will change with time. If you know the time delay of your output signal, you might be able to calculate the phase of a sinusoid at some point in the future. If you have a predicted phase of a particular frequency of noise at some time and location, you can add π to that phase angle to estimate the noice-cancelling signal.
If you don't know the frequency response and delay of your system, then you won't know what frequencies, amplitudes or phases of signal to create for cancellation. You might well end up amplifying the noise instead of cancelling it.
It seems that what’s missing is the propagation delay required to intercept and negate a signal. The KISS rule will eventually prove this true. The FFT is a complex calculation and each N iteration will introduce resulting error due to the time required to process the signal. To cancel a sound wave it will need to be intercepted in advance, processed and inverted. Then the time constant of the transducer must. E considered. My experience is that a microphone near the source of “noise” connected by wire and amplification device and transducer near the location where It is to be cancelled.
edit: typo
The basic idea of ANC is to find repetitive sound and play the opposite of it. If the repetitive sound continue to play we'll be able to cancel it. That goes in direct contradiction to to the other answers, but I'll clarify.
Playing the opposite sound means playing it again with a precise power and delay, possibly inverting the waveform. The delay itself varies for each frequency. For example, for a 20Hz sound we have to replay the inverted sound on a precise multiple of 1/20 = 0.05s. For 23Hz, for example, the delay has to be a multiple of 1/23 ~= 0.04347s.
Since any waveform can be produced by sum of sinusoidal, one way of doing it would be to only worry about the N biggest sinusoids, measured in power (square of the amplitudes). For finding the sinusoidal's frequencies and power we use the Fourier Transform, typically with the FFT algorithm.
If we take, for example N=8, it means we are trying to eliminate the 8 most powerfull wave components. For each of them we store:
wave's amplitude
wave's offset, taking the computer's clock as a base.
than we constantly play 8 sinusoids, each on the correct power and with the correct delay. The hard part is what happens next. We need to keep listening to adapt, but now we are listening to the environment sound + our own sound. This algorithm is harder to implement, but conceptually is easier, and one could easily figure out how to do it by himself.
So, contrary to what the other answers say, managing the time delay is critical. Is not possible to create an ANC system without doing it. If you only care about the frequency domain, the only thing you could possibly do is filter those frequencies. On an ANC system this makes not sense.