I would like to understand which is the maximum transmission range (maximum distance beetween two vehicles exchanging messages) having set the SimpleObstacleShadowing analogue model.
So I read this article and I would like to exploit formula (6) to reach my purpose. Unlikely I didn't understand if the attenuation got by obstacles (Lobs[dB]) is given in Veins or through which parameters I can actually compute it.
I also read this post but I think those formulas do not hold for this model.
Thanks so much
The maximum distance between two nodes in the network is calculated by the SimplePathLoss model. It calculates the distance according to the used transmission frequency and MCS.
The SimpleObstacleShadowing additionally adds attenuation if the transmitted message hits an obstacle.
Both are analogue models and can be configured via the config.xml which is read by the Decider module of the physical layer.
Update:
Also there is the maxInterfDist parameter which determines the distance an ideal communication can have at most.
Related
How to calculate the RSSI value in Veins, the calculation method and theoretical formula look different, find the codes as follows, but did not understand, looking forward to help.
double recvPower_dBm = 10 * log10(s.getAtCenterFrequency());
I don't know what you are assuming is the "theoretical formula" to calculate RSSI. I am assuming you are asking about how Veins calculates received power based on transmit power of a signal. As of Veins 5.1, the calculations roughly follow a typical link budget equation, taking into account transmit power, antenna gains, as well as various loss effects. One of these is path loss; the most simple path loss model, free space path loss, is being modeled by its SimplePathlossModel: Here, the attenuation is computed as , which (for alpha=2) mirrors the formula of free space path loss.
I have used to below hyper parameters to train the model.
rcf.set_hyperparameters(
num_samples_per_tree=200,
num_trees=250,
feature_dim=1,
eval_metrics =["accuracy", "precision_recall_fscore"])
is there any best way to choose the num_samples_per_tree and num_trees parameters.
what are the best numbers for both num_samples_per_tree and num_trees.
There are natural interpretations for these two hyper-parameters that can help you determine good starting approximations for HPO:
num_samples_per_tree -- the reciprocal of this value approximates the density of anomalies in your data set/stream. For example, if you set this to 200 then the assumption is that approximately 0.5% of the data is anomalous. Try exploring your dataset to make an educated estimate.
num_trees -- the more trees in your RCF model the less noise in scores. That is, if more trees are reporting that the input inference point is an anomaly then the point is much more likely to be an anomaly than if few trees suggest so.
The total number of points sampled from the input dataset is equal to num_samples_per_tree * num_trees. You should make sure that the input training set is at least this size.
(Disclosure - I helped create SageMaker Random Cut Forest)
I notice that resonance audio envelope the listener by HOA, I am curious about how do you deal with forbiden frequency (non-uniqueness) in HOA?
HOA describes sound fields accurately only up to a certain frequency and only within a limited region. This is because the spherical Bessel functions only vanish for small wavenumber * radius values. For higher values, indeed, they can be null and the sound field reconstruction will suffer from spatial aliasing errors.
To deal with this problem, I think it is a common practice to apply frequency-dependent smoothing functions in the spherical harmonics domain. For example, instead of trying to reconstruct a high-frequency sound field, we can apply 'MaxRe' weighting factors to form a different kind of an ambisonic 'panner', which will assure best energy concentration in the direction of the virtual source.
100 periods have been collected from a 3 dimensional periodic signal. The wavelength slightly varies. The noise of the wavelength follows Gaussian distribution with zero mean. A good estimate of the wavelength is known, that is not an issue here. The noise of the amplitude may not be Gaussian and may be contaminated with outliers.
How can I compute a single period that approximates 'best' all of the collected 100 periods?
Time-series, ARMA, ARIMA, Kalman Filter, autoregression and autocorrelation seem to be keywords here.
UPDATE 1: I have no idea how time-series models work. Are they prepared for varying wavelengths? Can they handle non-smooth true signals? If a time-series model is fitted, can I compute a 'best estimate' for a single period? How?
UPDATE 2: A related question is this. Speed is not an issue in my case. Processing is done off-line, after all periods have been collected.
Origin of the problem: I am measuring acceleration during human steps at 200 Hz. After that I am trying to double integrate the data to get the vertical displacement of the center of gravity. Of course the noise introduces a HUGE error when you integrate twice. I would like to exploit periodicity to reduce this noise. Here is a crude graph of the actual data (y: acceleration in g, x: time in second) of 6 steps corresponding to 3 periods (1 left and 1 right step is a period):
My interest is now purely theoretical, as http://jap.physiology.org/content/39/1/174.abstract gives a pretty good recipe what to do.
We have used wavelets for noise suppression with similar signal measured from cows during walking.
I'm don't think the noise is so much of a problem here and the biggest peaks represent actual changes in the acceleration during walking.
I suppose that the angle of the leg and thus accelerometer changes during your experiment and you need to account for that in order to calculate the distance i.e you need to know what is the orientation of the accelerometer in each time step. See e.g this technical note for one to account for angle.
If you need get accurate measures of the position the best solution would be to get an accelerometer with a magnetometer, which also measures orientation. Something like this should work: http://www.sparkfun.com/products/10321.
EDIT: I have looked into this a bit more in the last few days because a similar project is in my to do list as well... We have not used gyros in the past, but we are doing so in the next project.
The inaccuracy in the positioning doesn't come from the white noise, but from the inaccuracy and drift of the gyro. And the error then accumulates very quickly due to the double integration. Intersense has a product called Navshoe, that addresses this problem by zeroing the error after each step (see this paper). And this is a good introduction to inertial navigation.
Periodic signal without noise has the following property:
f(a) = f(a+k), where k is the wavelength.
Next bit of information that is needed is that your signal is composed of separate samples. Every bit of information you've collected are based on samples, which are values of f() function. From 100 samples, you can get the mean value:
1/n * sum(s_i), where i is in range [0..n-1] and n = 100.
This needs to be done for every dimension of your data. If you use 3d data, it will be applied 3 times. Result would be (x,y,z) points. You can find value of s_i from the periodic signal equation simply by doing
s_i(a).x = f(a+k*i).x
s_i(a).y = f(a+k*i).y
s_i(a).z = f(a+k*i).z
If the wavelength is not accurate, this will give you additional source of error or you'll need to adjust it to match the real wavelength of each period. Since
k*i = k+k+...+k
if the wavelength varies, you'll need to use
k_1+k_2+k_3+...+k_i
instead of k*i.
Unfortunately with errors in wavelength, there will be big problems keeping this k_1..k_i chain in sync with the actual data. You'd actually need to know how to regognize the starting position of each period from your actual data. Possibly need to mark them by hand.
Now, all the mean values you calculated would be functions like this:
m(a) :: R->(x,y,z)
Now this is a curve in 3d space. More complex error models will be left as an excersize for the reader.
If you have a copy of Curve Fitting Toolbox, localized regression might be a good choice.
Curve Fitting Toolbox supports both lowess and loess localized regression models for curve and curve fitting.
There is an option for robust localized regression
The following blog post shows how to use cross validation to estimate an optimzal spaning parameter for a localized regression model, as well as techniques to estimate confidence intervals using a bootstrap.
http://blogs.mathworks.com/loren/2011/01/13/data-driven-fitting/
First I am going to broadly state what I'm trying to do and ask for advice. Then I will explain my current approach and ask for answers to my current problems.
Problem
I have an MP3 file of a person speaking. I'd like to split it up into segments roughly corresponding to a sentence or phrase. (I'd do it manually, but we are talking hours of data.)
If you have advice on how to do this programatically or for some existing utilities, I'd love to hear it. (I'm aware of voice activity detection and I've looked into it a bit, but I didn't see any freely available utilities.)
Current Approach
I thought the simplest thing would be to scan the MP3 at certain intervals and identify places where the average volume was below some threshold. Then I would use some existing utility to cut up the mp3 at those locations.
I've been playing around with pymad and I believe that I've successfully extracted the PCM (pulse code modulation) data for each frame of the mp3. Now I am stuck because I can't really seem to wrap my head around how the PCM data translates to relative volume. I'm also aware of other complicating factors like multiple channels, big endian vs little, etc.
Advice on how to map a group of pcm samples to relative volume would be key.
Thanks!
PCM is a time frame base encoding of sound. For each time frame, you get a peak level. (If you want a physical reference for this: The peak level corresponds to the distance the microphone membrane was moved out of it's resting position at that given time.)
Let's forget that PCM can uses unsigned values for 8 bit samples, and focus on
signed values. If the value is > 0, the membrane was on one side of it's resting position, if it is < 0 it was on the other side. The bigger the dislocation from rest (no matter to which side), the louder the sound.
Most voice classification methods start with one very simple step: They compare the peak level to a threshold level. If the peak level is below the threshold, the sound is considered background noise.
Looking at the parameters in Audacity's Silence Finder, the silence level should be that threshold. The next parameter, Minimum silence duration, is obviously the length of a silence period that is required to mark a break (or in your case, the end of a sentence).
If you want to code a similar tool yourself, I recommend the following approach:
Divide your sound sample in discrete sets of a specific duration. I would start with 1/10, 1/20 or 1/100 of a second.
For each of these sets, compute the maximum peak level
Compare this maximum peak to a threshold (the silence level in Audacity). The threshold is something you have to determine yourself, based on the specifics of your sound sample (loudnes, background noise etc). If the max peak is below your threshold, this set is silence.
Now analyse the series of classified sets: Calculate the length of silence in your recording. (length = number of silent sets * length of a set). If it is above your Minimum silence duration, assume that you have the end of a sentence here.
The main point in coding this yourself instead of continuing to use Audacity is that you can improve your classification by using advanced analysis methods. One very simple metric you can apply is called zero crossing rate, it just counts how often the sign switches in your given set of peak levels (i.e. your values cross the 0 line). There are many more, all of them more complex, but it may be worth the effort. Have a look at discrete cosine transformations for example...
Just wanted to update this. I'm having moderate success using Audacity's Silence Finder. However, I'm still interested in this problem. Thanks.
PCM is a way of encoding a sinusoidal wave. It will be encoded as a series of bits, where one of the bits (1, I'd guess) indicates an increase in the function, and 0 indicates a decrease. The function can stay roughly constant by alternating 1 and 0.
To estimate amplitude, plot the sin wave, then normalize it over the x axis. Then, you should be able to estimate the amplitude of the sin wave at different points. Once you've done that, you should be able to pick out the spots where amplitude is lower.
You may also try to use a Fourier transform to estimate where the signals are most distinct.