The Unetstack handbook refers to Urick's classic book 'The Principles of Sound', pp 105-111, 3rd Ed for transmission loss. There are many causes for absorption described therein. What is accounted for when estimating the absorption in Unetstack?
Second question, how do interference/collisions work in the Basic Acoustic Model, is interference from other nodes treated as added noise or something in that direction?
Losses taken into account in BasicAcousticChannel model:
Spreading loss
Frequency-dependent absorption (based on Schulkin & Marsh)
The BasicAcousticChannel model treats interference as noise and computes a SINR. The SINR is used to compute the detection probability and the BER for a BPSK modulation with Rician fading and a processing/coding gain. The parameters of the model (e.g. ricianK, pfa, processingGain, fastFading, noiseLevel, bandwidth, spreading, carrierFrequency, etc) control the details of the gain, fading, etc.
Related
I have to use a uninformed search technique to solve the following problem.
The game is like:
On side of the river, there is a Policeman, a Robber, a woman in a red-dress and her two children, a woman in a yellow dress and her two children. There is a boat that can carry atmost two persons. The children cannot drive the boat.
If the policeman is absent then the robber will kill the people. If the red-dress woman is absent then the yellow-dressed woman will kill the red-dressed woman’s children and vice versa.
I am confused as usual. Please help me figure it out.
The problem and how can it be solved (without programming) is shown in the video below:
https://www.youtube.com/watch?v=vSusAZBSWwg
Thank you.
Problems like River Crossing Puzzle, Sokoban or Lemmings are solved normally with Brute-Force-Search in the gametree. The domain is specified declarative as rules (moves are possible or not), and a function which determines the amount of points which are reached by a policy (policy = plan through the gametree). The solver has the aim to find a good policy. The best hardware for doing this is a quantum computer with unlimited speed for testing as much as possble moves per second.
The reason why this is not practicaly is because of a phenomenon which is called "combinatorial explosion", first introduced by James Lighthill in year 1973 for prooving that artifical intelligence is not ready for use in realworld. The answer to that problem is, to use alternative strategies which have noting to do with brute-force-search.
One possibility is to use heuristics which are hardcoded into programcode. Often these heuristics are called macroaction or motion primitives. An example would be "bring-robber-to-other-side". This subfunction executes a predefined number of actions. Another macroaction could be "check-if-two-woman-are-on-the-same-side". To implement such kind of strategy for the complete game is a hard task. Not because of high cpu usage, but because of every detail has to coded into software.
I'm developing an IOS app for frequency detection, and I'm using the YIN algorithm, which is very precise: witch Audacity, I've generated rectangular waves of different frequencies - and my algorithm has a precision of about 0.1 % - for example generating a tone of 82,4 Hz (E string), I really get 82,4 Hz and nothing else.
Anyhow, when I strum a guitar string, I often get overtones which sometimes can be stronger (with a higher amplitude) than the fundamental tone (F0). Consequently, my display starts "dancing" and toggling - sometimes, it even occurs that (when the tone dies out) my algorithm stops at the overtone's frequency (for example A instead of E) - so the user has to strum the string again in oder to see if his desired tone (frequency) is present.
I know that this phenomena has nothing to do with my algorithm, because it's merely a "hardware" problem (I mean the guitar which simply produces overtones).
I've tried in vain to smooth the results (of the frequency detection) or to "snap" to a fixed frequency as soon as a crucial frequency (for example 82.4 Hz for E string +/- tolerance) has been detected. Anyhow, it often occurrs that my algorithm snaps into an erroneous frequency, as well.
I'm asking myself how cheap guitar tuners (for 10$ in guitar stores) are working, as their frequency detections are reliable and stable, as well.
I don't want to change the algorithm, but two possible solutions come into my mind:
Preprocessing of the signal (maybe Hanning window, lowpass or bandpass filtering) and/or
Postprocessing of the signal (some kind of frequency smoothing).
Has someone an idea how to overcome the "choppy" results?
I used autocorrelation for my free chromatic app iTransposer and incorporated a Hanning window so this may help you. I wasn't looking for accuracy initially as I wanted to display the note on a stave not a meter. However a friend of mine tested it to 0.1 Hz with a signal generator at his work and had issues over 383 Hz with simple signals such as Sine waves.I've tried it with various brass instruments, guitar and Garageband instruments seems to be OK for tuning.
Basically I implemented this http://www.ucl.ac.uk/~ucjt465/tutorials/praatpitch.html
using VDSP and updated a sample project supplied by Kevin P Murphy https://github.com/kevmdev/PitchDetectorExample
I have a problem with recognising a signal. Let say the signal is a quasiperiodic signal, the period time has finite limits. The "shape" of the signal must match some criteria, so the actual algorithm using signal processing techniques such as filtering, derivating the the signal, looking for maximum and minimum values. It has a good rate at finding the good signals, but the problem is it also detects wrong shapes too.
So I want to use Aritifical Intelligence - mainly Neural Networks - to overcome this problem. I thought that a multi layer network with some average inputs (the signal can be reduced) and one output whould shows the "matching" from 0..1. However the problem is that I never did such a thing, so I am asking for help, how to achive something like this? How to teach the neural network to get the expected results? (let say I have vectors for inputs which should give 1 as output)
Or this whole idea is a wrong approximation for the problem? I am open to any learning algorithms or idea to learn and use to overcome on this problem.
So here is a figure on the measured signal(s) (values and time is not a concern now) and you can see a lot "wrong" ones, the most detected signals are good as mentioned above.
Your question can be answered in a broad manner. You should consider editing it to prevent it to be closed.
But anyway, Matlab had a lot of built-in function and toolbox to support Artificial Intelligence, with a lot of sample code available, which you can modify and refer to. You can find some in Matlab FileExchange.
And I know reading a lot of technical paper for Artificial Intelligence is a daunting task, so good luck!
You can try to build a neural network using Neuroph. You can inspire from "http://neuroph.sourceforge.net/TimeSeriesPredictionTutorial.html".
On the other hand, it is possible to approximate the signal using Fourier transformation.
You can try 1D convolution. So the basic idea is you give a label 0: bad, 1: good to each signal value at each timestamp. After this, you can model
model = Sequential()
model.add(Conv1D(filters=64, kernel_size=3, activation='relu', padding = 'same', input_shape=(1,1)))
model.add(Bidirectional(LSTM(20, return_sequences=True)))
model.add(Conv1D(filters=64, kernel_size=3, activation='relu', padding = 'same'))
model.add(Dropout(0.5))
model.add(Flatten())
model.add(Dense(100, activation='sigmoid'))
model.add(Dense(2, activation='softmax'))
model.compile(loss='binary_crossentropy', optimizer='adam', metrics=['accuracy'])
Train the model and then give it a new signal to predict. It will predict given series to 0 and 1 values. if count of 0 is more than count of 1, the signal is not good.
I have a video which has got turn left,turn right etc marks on the roads.
I have to detect those signs.I am going ahead with template matching in which I am matching the edge detected outputs,But I am not getting satisfactory results,Is there any other way to detect it? Please help.
If you want a solution that is not too complicated but more robust than template matching, I suggest you'd go for Hough voting on SIFT descriptors. This method is provides some degree of robustness to various problems, including partial occlusion of the sign, illumination variations and deformations of the sign. In particular, the method is completely invariant to rotation and uniform scaling of the template object.
The basic idea of the algorithm is as follows:
a) extract SIFT features from the template and query images.
b) set an arbitrary reference point in the template image and calculate, for each keypoint in the template image, the vector from the keypoint to the reference point.
c) match keypoints from the template image to the query image.
d) cast a vote for each matched keypoint for all object locations in the query image that this keypoint agrees with. You do that using the vectors calculated in step (b) and the location, scale and orientation of the matched keypoints in the query image.
e) If the object is indeed located in the image, the votes map should have a strong local maximum at it's location.
f) Optionally, you can verify the detection by using template matching.
You can read more about that method on Wikipedia here or in the original paper (by D. Lowe) here.
Using SIFT or SURF. You can get the invariable descriptor with training you can determine if the vector that represent the road marks (turn left, right or stop) match with the new in the video.
You might try extracting features and training a classifier (linear discriminant, neural network, naive Bayes, etc.). There are many candidate features you might try, but I'd think that you wouldn't need anything too complicated, even if the edge detection is poor, assuming that isolation of the sign is good. Some features to consider are: horizontal and vertical projections (row and column totals) and simple statistics of edge pixels (mean, standard deviation, skewness, etc. For more feature ideas, see any of these books:
"Shape Classification and Analysis: Theory and Practice", by Costa and Cesar
"Algorithms for Image Processing and Computer Vision", by J. R. Parker
"Digital Image Processing", by Gonzalez and Woods
Not unlike a clap detector ("Clap on! clap clap Clap off! clap clap Clap on, clap off, the Clapper! clap clap ") I need to detect when a door closes. This is in a vehicle, which is easier than a room or household door:
Listen: http://ubasics.com/so/van_driver_door_closing.wav
Look:
It's sampling at 16bits 4khz, and I'd like to avoid lots of processing or storage of samples.
When you look at it in audacity or another waveform tool it's quite distinctive, and almost always clips due to the increase in sound pressure in the vehicle - even when the windows and other doors are open:
Listen: http://ubasics.com/so/van_driverdoorclosing_slidingdoorsopen_windowsopen_engineon.wav
Look:
I expect there's a relatively simple algorithm that would take readings at 4kHz, 8 bits, and keep track of the 'steady state'. When the algorithm detects a significant increase in the sound level it would mark the spot.
What are your thoughts?
How would you detect this event?
Are there code examples of sound pressure level calculations that might help?
Can I get away with less frequent sampling (1kHz or even slower?)
Update: Playing with Octave (open source numerical analysis - similar to Matlab) and seeing if the root mean square will give me what I need (which results in something very similar to the SPL)
Update2: Computing the RMS finds the door close easily in the simple case:
Now I just need to look at the difficult cases (radio on, heat/air on high, etc). The CFAR looks really interesting - I know I'm going to have to use an adaptive algorithm, and CFAR certainly fits the bill.
-Adam
Looking at the screenshots of the source audio files, one simple way to detect a change in sound level would be to do a numerical integration of the samples to find out the "energy" of the wave at a specific time.
A rough algorithm would be:
Divide the samples up into sections
Calculate the energy of each section
Take the ratio of the energies between the previous window and the current window
If the ratio exceeds some threshold, determine that there was a sudden loud noise.
Pseudocode
samples = load_audio_samples() // Array containing audio samples
WINDOW_SIZE = 1000 // Sample window of 1000 samples (example)
for (i = 0; i < samples.length; i += WINDOW_SIZE):
// Perform a numerical integration of the current window using simple
// addition of current sample to a sum.
for (j = 0; j < WINDOW_SIZE; j++):
energy += samples[i+j]
// Take ratio of energies of last window and current window, and see
// if there is a big difference in the energies. If so, there is a
// sudden loud noise.
if (energy / last_energy > THRESHOLD):
sudden_sound_detected()
last_energy = energy
energy = 0;
I should add a disclaimer that I haven't tried this.
This way should be possible to be performed without having the samples all recorded first. As long as there is buffer of some length (WINDOW_SIZE in the example), a numerical integration can be performed to calculate the energy of the section of sound. This does mean however, that there will be a delay in the processing, dependent on the length of the WINDOW_SIZE. Determining a good length for a section of sound is another concern.
How to Split into Sections
In the first audio file, it appears that the duration of the sound of the door closing is 0.25 seconds, so the window used for numerical integration should probably be at most half of that, or even more like a tenth, so the difference between the silence and sudden sound can be noticed, even if the window is overlapping between the silent section and the noise section.
For example, if the integration window was 0.5 seconds, and the first window was covering the 0.25 seconds of silence and 0.25 seconds of door closing, and the second window was covering 0.25 seconds of door closing and 0.25 seconds of silence, it may appear that the two sections of sound has the same level of noise, therefore, not triggering the sound detection. I imagine having a short window would alleviate this problem somewhat.
However, having a window that is too short will mean that the rise in the sound may not fully fit into one window, and it may apppear that there is little difference in energy between the adjacent sections, which can cause the sound to be missed.
I believe the WINDOW_SIZE and THRESHOLD are both going to have to be determined empirically for the sound which is going to be detected.
For the sake of determining how many samples that this algorithm will need to keep in memory, let's say, the WINDOW_SIZE is 1/10 of the sound of the door closing, which is about 0.025 second. At a sampling rate of 4 kHz, that is 100 samples. That seems to be not too much of a memory requirement. Using 16-bit samples that's 200 bytes.
Advantages / Disadvantages
The advantage of this method is that processing can be performed with simple integer arithmetic if the source audio is fed in as integers. The catch is, as mentioned already, that real-time processing will have a delay, depending on the size of the section that is integrated.
There are a couple of problems that I can think of to this approach:
If the background noise is too loud, the difference in energy between the background noise and the door closing will not be easily distinguished, and it may not be able to detect the door closing.
Any abrupt noise, such as a clap, could be regarded as the door is closing.
Perhaps, combining the suggestions in the other answers, such as trying to analyze the frequency signature of the door closing using Fourier analysis, which would require more processing but would make it less prone to error.
It's probably going to take some experimentation before finding a way to solve this problem.
You should tap in to the door close switches in the car.
Trying to do this with sound analysis is overengineering.
There are a lot of suggestions about different signal processing
approaches to take, but really, by the time you learn about detection
theory, build an embedded signal processing board, learn the processing
architecture for the chip you chose, attempt an algorithm, debug it, and then
tune it for the car you want to use it on (and then re-tune and re-debug
it for every other car), you will be wishing you just stickey taped a reed
switch inside the car and hotglued a magnet to the door.
Not that it's not an interesting problem to solve for the dsp experts,
but from the way you're asking this question, it's clear that sound
processing isn't the route you want to take. It will just be such a nightmare
to make it work right.
Also, the clapper is just an high pass filter fed into a threshold detector. (plus a timer to make sure 2 claps quickly enough together)
There is a lot of relevant literature on this problem in the radar world (it's called detection theory).
You might have a look at "cell averaging CFAR" (constant false alarm rate) detection. Wikipedia has a little bit here. Your idea is very similar to this, and it should work! :)
Good luck!
I would start by looking at the spectral. I did this on the two audio files you gave, and there does seem to be some similarity you could use. For example the main difference between the two seems to be around 40-50Hz. My .02.
UPDATE
I had another idea after posting this. If you can, add an accelerometer onto the device. Then correlate the vibrational and acoustic signals. This should help with cross vehicle door detection. I'm thinking it should be well correlated since the sound is vibrationally driven, wheres the stereo for example, is not. I've had a device that was able to detect my engine rpm with a windshield mount (suction cup), so the sensitivity might be there. (I make no promises this works!)
(source: charlesrcook.com)
%% Test Script (Matlab)
clear
hold all %keep plots open
dt=.001
%% Van driver door
data = wavread('van_driver_door_closing.wav');
%Frequency analysis
NFFT = 2^nextpow2(length(data));
Y = fft(data(:,2), NFFT)/length(data);
freq = (1/dt)/2*linspace(0,1,NFFT/2);
spectral = [freq' 2*abs(Y(1:NFFT/2))];
plot(spectral(:,1),spectral(:,2))
%% Repeat for van sliding door
data = wavread('van_driverdoorclosing.wav');
%Frequency analysis
NFFT = 2^nextpow2(length(data));
Y = fft(data(:,2), NFFT)/length(data);
freq = (1/dt)/2*linspace(0,1,NFFT/2);
spectral = [freq' 2*abs(Y(1:NFFT/2))];
plot(spectral(:,1),spectral(:,2))
The process for finding distinct spike in audio signals is called transient detection. Applications like Sony's Acid and Ableton Live use transient detection to find the beats in music for doing beat matching.
The distinct spike you see in the waveform above is called a transient, and there are several good algorithms for detecting it. The paper Transient detection and classification in energy matters describes 3 methods for doing this.
I would imagine that the frequency and amplitude would also vary significantly from vehicle to vehicle. Best way to determine that would be taking a sample in a Civic versus a big SUV. Perhaps you could have the user close the door in a "learning" mode to get the amplitude and frequency signature. Then you could use that to compare when in usage mode.
You could also consider using Fourier analysis to eliminate background noises that aren't associated with the door close.
Maybe you should try to detect significant instant rise in air pressure that should mark a door close. You can pair it with this waveform and sound level analysis and these all might give you a better result.
On the issue of less frequent sampling, the highest sound frequency which can be captured is half of the sampling rate. Thus, if the car door sound was strongest at 1000Hz (for example) then a sampling rate below 2000Hz would lose that sound entirely
A very simple noise gate would probably do just fine in your situation. Simply wait for the first sample whose amplitude is above a specified threshold value (to avoid triggering with background noise). You would only need to get more complicated than this if you need to distinguish between different types of noise (e.g. a door closing versus a hand clap).