So i'm trying to figure out how much Y a player has fallen in a game to then figure out how much damage should be taken. I'm doing this by trying to store the current y position of a player when they aren't on the ground, getting the new y position then subtracting the new position with the old one. Problem is oldy is always set to the current y and i don't know how to keep it separate. I don't have a lot of experience with C so any help would be appreciated
if (player->grounded==false){
player->blocksfallen=player->position.y-player->oldy;
player->oldy=player->position.y;
} else {
player->blocksfallen=0;
}
The damage should depend on the speed with which the ground is hit, not by the height fallen, though there is a relation between those.
The relation is not 1:1, consider somebody starting to fall with already a downward speed, e.g. being thrown downwards. Or the other direction, the speed might be slowed by something like a parachute.
Inside the code this translates to having a variable with (at least the vertical component of) the speed. Increase it for acceleration, decrease it for any breaking effects (more precisely phrased: increase it less for breaking effects).
When you hit the ground, calculate damage based on the speed at that point.
Introducing that speed variable will also solve your problem in the absence of thrown characters and parachutes. It is then something of an accumulator of accelearation, which is very close to an accumulator of distance fallen - just easier to update. You just need to increase it by the gravity-caused acceleration each turn/frame/animation-step.
This means to turn the calculation around. Instead of trying to calculate the speed/total-height/damage from the previous height andcurrent height, you calculate the current height from previous height + speed. If you look at how you calculate the current height form previous you will notice that you already do something similar there (let's say I am pretty sure you do, though it is not seen in the shown code, something like height=height-speedconstant).
I am using Appium's java-client 3.2.0 API; and would like to perform a "flick" gesture. I am guessing I need to use the swipe() method for this. My questions are
What value should I use as the duration so that the swipe is "fast" enough to be considered as a flick gesture?
Does the distance between start and end point matter?
if duration is fixed, longer distance means faster flick speed;
if distance is fixed, shorter duration means faster flick speed.
You can always set a breakpoint at the flicking point and change different values to see whether it's 'fast' enough.
Now I am trying to implement a particle filter. I am given a wall-mounted map, and I try to localize a robot in this map. Based on particle filter method, I initialize 1000 random particles, and in each step, I move these 1000 particles according to a certain movement instruction, i.e. an angle-odometry pair. After a move, I calculate the likelihood of the measurements compared to the sensed distance to the wall, and then resample the particles based on their likelihoods. I think this is the basic process for particle filter. What confuses me now is that how should I deal with the situations where some of the particles hit the wall while they are forwarding?
I think it is too late for you. However, it may help other people. Particle filter is a probabilistic approach, where particles can be sampled everywhere based on motion and prior distributions.
In your case, you can sample on the wall without any worry. Afterwards, the likelihood process will return a very low probability for that particle and it will be automatically resampled to another one with higher probability.
I have created a very simple numerical simulation that models an object being thrown off a building at some angle, and when the object hits the ground, the simulation stops. Now I want to add in collision detection. How would I go about doing this?
I know I need to find the exact time that the object (a ball) hits the ground, as well as the velocity in the x and y direction, and position of the object when it hits the ground, and I have to add in parameters that say how much the ball will bounce on impact. But I don't know how to go about doing this. I know that there are various ways of detecting collision but since I am new to this, the most comprehensible method would be best.
Make a coordinate system, with the ground at y=0. Track the coordinates of the ball as it flies and then check when it has y=0, and that's where it hits the ground. You can also keep track of the x and y velocity as the ball is moving.
Use Physics skillz. This is a good tutorial. If you have it, I recommend Fundamentals of Physics by Halliday, Resnick and Walker. They have a very good chapter on this.
If you are just looking for the math, that you could write C code for. I found this one helpful. Math Models
Collision detection simply involves determining the distance between 2 objects.
If you are only interested in collisions between objects and the ground, you can use:
if(object.y <= ground.y) {
//collision occurred
}
To do collisions between objects, you can loop through all objects and compare them to each other in the same way.
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).