how to convert RSSI to distance?. i have three coordinate RFID reader : Reader1(0,0). reader2(0,8), reader3(8,5). Transponder get RSSI, 156 from reader1, 115 from reader2 and 124 from reader3. how i can get distance between Transponder to reader1, reader2 and reader3? also coordinate of Transponder.
Signal strength from RFID Reader is 0-255. 255 if so close, 0 if so far.
i got RSSI formula from internet, but i confused to translate the parameters of A and NAi.
can u help me?
thx u :-)
Buddy, let me clarify it: RSSI (received signal strength indicator) do not relates with the distance. We must understand that the RSSI value is given in decibel, a logarithm relation with the power ratio expressed in watt. The formula is Db = 10*Log(Watt).
And why it does not relates with distance? Its because a signal strength ratio depends on the environment (such walls, deep, structures, and so on) that makes the signal propagate more easily or more hardly. In other words, you can have different RSSI values for the same distance, as we can experiment if we use a cellphone at an open area, or inside a road tunnel.
Again, RSSI is not related with distance. The only situation where you can freely convert these two values is in theoretical non-practical situations, where environment (earth form, structures, air density, etc) do not takes relevance. In these situations, you fall in physics context, and not in stackoverflow (programming) context.
Related
This might sound like a very silly question, so I apologize if this is something very simple but I just cannot get my head around it. I am trying to understand what the data provides in terms of real time information, for example, the MPU-6050:
Gyroscope - is a 16 bit data register with a range from (0 <-> 65535)
There is a selection of ranges (±250, ±500, ±1000, and ±2000°/sec)
If the range is set to ±250°/sec, is the reading 360/65535 = 0.0054 resolution?
What does °/sec mean, if the sensor does not move and reads zero and then turned quickly does it mean it will be reading the angle at the set range? For example, if the range was set to ±2000°/sec and it was moved 200° would the read move from 0 to (2/65535 *200) and keep sending this value once the sensor stopped moving?
Accelerometer - is a 16 bit data register with a range from (0 <-> 65535)
There is a selection of ranges (±2g, ±4g, ±8g and ±16g)
If the sensor is not moving, completely flat the reading will be 0?
If the sensor is shocked at 2g will the max reading be 65535 (if set of 2g, with a resolution of 2/65535)
If the sensor is shocked at 16g will the max reading be 65535 (if set of 16g, with a resolution of 16/65535))
There are two main documents regarding the MPU6050, and those are the datasheet and the register map.
The gyro measurements are stored in the GYRO_XOUT, GYRO_YOUT, GYRO_ZOUT parameters, as you can see in the register map document, page 31. Each parameter is stored as a 2-complement signed 16 bit value split into two 8-bit registers: the GYRO_xOUT_L and _H.
In the same page, you can see the sensitivity for each full-scale range. For example, if your FSR is +/- 250º/sec, and you want to measure 1º/sec, the GYRO_xOUT parameter should read 131 counts.
The accelerometer-related registers can be seen in the same document, page 29. The idea is the same, two 8-bit registers to form a 2-complement signed 16-bit value, and the sensitivity values for each FSR.
Regarding your question in comments, if you rotate the device 125º in a second, at constant rotation speed, you should read 16375 in the rotation registers during the movement. This value comes from 131 counts/(º/sec) * 125º/sec = 16375 counts.
I am doing a project on self balancing quadcopter with Autonomous control. I am using Arduino Mega 2560 and MPU6050. I have obtained the roll and pitch angles from MPU6050 without the help of DMP and applied complex filter to omit the noise due to vibration.
Also configured and able to run the BLDC motors with Flysky Transmitter and receiver with the help of Arduino interrupts. Now for balancing I am focusing on only one axis (i.e. roll). I have also constructed a balancing stand for the free movement of roll axis by the motor.
For the controlling part, I am implementing PID algorithm. I tried using only the kp value so that, somehow I can balance and then move on to ki and kd term. But unfortunately, for Kp itself, the quadcopter is undergoing aggressive oscillation and is not settling at all.
Some of my queries are:
Whether a single PID loop is enough, or we have to add another?
What type of tuning method I can implement, to find the kp, ki, kd other than trial and error?
I programmed my ESC for 1000 to 2000 microseconds. My PID input angles will be within the range +/- 180. Whether I can directly set the PID output limits for range -1000 to 1000 or -180 to 180 or any other value?
The code can read from the URL https://github.com/antonkewin/quadcopter/blob/master/quadpid.ino
Since its not provided, I am assuming that:
The Loop time is atleast 4ms. (The less the better)
The sensor noise is been reduced to an acceptable level.
MPU-6050 needs gyro+accel data to be combined to get angles in degrees.
If the above points are not taken care of, it will Not balance itself.
Initially, you can get away without tuning kI. So let's focus on kP and kD:
Keep increasing Kp till it starts oscillate fast. Keep the Kp value half of that.
With kP set, start experimenting kD values, as it will try to dampen the overshoots of kP.
Fiddle around these two values, tune it for perfection.
Note, the more accurate your gyro data is, the higher you can set your kP to.
Im really confused over here. I am a ai programmer working on a game that is designed to detect beats in songs and some more. I have no previous knowledge about audio and just reading through whatever material i can find. While i got fft working and stuff I simply don't understand the way samples are transferred to different frequencies. Question 1, what does each frequency stands for. For the algorithm i got. I can transfer for example 1024 samples into 512 outcomes. So are they a description of the strength of each spectrum at the current second? it doesn't really make sense since what i remember is that there are 20,000hz in a 44.1khz audio recording. So how does 512 spectrum samples explain what is happening in that moment? Question 2, from what i read, its a number that represent the sound wave at this moment. However i read that by squaring both left channel and right channel, and add them together and you will get the current power level. Both these seems incoherent to my understanding, and i am really buff led so please explain away.
DFT output
the output is complex representation of phasor (Re,Im,Frequency) of basis function (usually sin wave). First item is DC offset so skip it. All the others are multiples of the same fundamental frequency (sampling rate/N). The output is symmetric (if the input is real only) so use just first half of results. Often power spectrum is used
Amplitude=sqrt(Re^2+Im^2)
which is the amplitude of basis function. If phase is needed then
phase=atan2(Im,Re)
beware DFT results are strongly dependent on the input signal shape,frequency and phase shift to your basis functions. That causes the output to vibrate/oscillate around the correct value and produce wide peaks instead of sharp ones for singular frequencies not to mention aliasing.
frequencies
if you got 44100Hz then the max output frequency is half of it that means the biggest frequency present in data is 22050Hz. The DFFT however does not contain this frequency so if you ignore the mirrored second half of results then:
for 4 samples DFT outputs frequencies are { -,11025 } Hz
for 8 samples frequencies are: { -,5512.5,11025,16537.5 } Hz
The output frequency is linear to its address from start so if you got N=512 samples
do DFFT on it
obtain first N/2=256 results
i-th sample represents frequency f=i*samplerate/N Hz
where i={ 1,...,(N/2)-1} ... skipping i=0
the image shows one of mine utility apps tighted together with
2-channel sound generator (top left)
2-channel oscilloscope (top right)
2-channel spectral analyzer (bottom) ... switched to linear frequency scale to make obvious what I mean in above text
zoom the image to see the settings ... I made it as close to the real devices as I could.
Here DCT and DFT comparison:
Here the DFT output dependency on input signal frequency aliasing by sampling rate
more channels
Summing power of channels is more safe. If you just add the channels then you could miss some data. For example let left channel is playing 1 Khz sin wave and the right exact opposite so if you just sum them then the result is zero but you can hear the sound .... (if you are not exactly in the middle between speakers). If you analyze each channel independently then you need to calculate DFFT for each channel but if you use power sum of channels (or abs sum) then you can obtain the frequencies for all channels at once , of coarse you need to scale the amplitudes ...
[Notes]
Bigger the N nicer the result (less aliasing artifacts and closer to the max frequency). For specific frequencies detection are FIR filter detectors more precise and faster.
Strongly recommend to read DFT and all sublinks there and also this plotting real time Data on (qwt) Oscillocope
I have a raw ecg signal, that contains complex values (real and imaginary) in time. Now I have to clear that signal out, remove noises, and flatten the signal.
The algorithm to do this that i know of is fast fourier transformation (FFT), but it doesnt flatten the signal, instead it generates correct fourier transformation, but the signal is not flat, it has high values on both sides. How can i do that?
I am doing this in java language, but I dont ask for the code, just for the hint with the idea, or an algorithm.
Thanks!
FFT doesn't flatten signal, it translates signal from time domain to frequency domain. If you signal is pure real, FT is symmetric - so you can see similar high peaks at both ends - this is very low frequency part of your signal.
To filter a signal, you can execute FT, apply some function to the result of transform - for example, lower high and very low frequency regions, and execute backward FT to return in the time domain.
I'm designing a real time Audio Analyser to be embedded on a FPGA chip. The finished system will read in a live audio stream and output frequency and amplitude pairs for the X most prevalent frequencies.
I've managed to implement the FFT so far, but it's current output is just the real and imaginary parts for each window, and what I want to know is, how do I convert this into the frequency and amplitude pairs?
I've been doing some reading on the FFT, and I see how they can be turned into a magnitude and phase relationship but I need a format that someone without a knowledge of complex mathematics could read!
Thanks
Thanks for these quick responses!
The output from the FFT I'm getting at the moment is a continuous stream of real and imaginary pairs. I'm not sure whether to break these up into packets of the same size as my input packets (64 values), and treat them as an array, or deal with them individually.
The sample rate, I have no problem with. As I configured the FFT myself, I know that it's running off the global clock of 50MHz. As for the Array Index (if the output is an array of course...), I have no idea.
If we say that the output is a series of One-Dimensional arrays of 64 complex values:
1) How do I find the array index [i]?
2) Will each array return a single frequency part, or a number of them?
Thankyou so much for all your help! I'd be lost without it.
Well, the bad news is, there's no way around needing to understand complex numbers. The good news is, just because they're called complex numbers doesn't mean they're, y'know, complicated. So first, check out the wikipedia page, and for an audio application I'd say, read down to about section 3.2, maybe skipping the section on square roots: http://en.wikipedia.org/wiki/Complex_number
What that's telling you is that if you have a complex number, a + bi, you can picture it as living in the x,y plane at location (a,b). To get the magnitude and phase, all you have to do is find two quantities:
The distance from the origin of the plane, which is the magnitude, and
The angle from the x-axis, which is the phase.
The magnitude is simple enough: sqrt(a^2 + b^2).
The phase is equally simple: atan2(b,a).
The FFT result will give you an array of complex values. The twice the magnitude (square root of sum of the complex components squared) of each array element is an amplitude. Or do a log magnitude if you want a dB scale. The array index will give you the center of the frequency bin with that amplitude. You need to know the sample rate and length to get the frequency of each array element or bin.
f[i] = i * sampleRate / fftLength
for the first half of the array (the other half is just duplicate information in the form of complex conjugates for real audio input).
The frequency of each FFT result bin may be different from any actual spectral frequencies present in the audio signal, due to windowing or so-called spectral leakage. Look up frequency estimation methods for the details.