Given an audio stream, find when a door slams (sound pressure level calculation?) - c

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).

Related

How can I get current microphone input level with C WinAPI?

Using Windows API, I want to implement something like following:
i.e. Getting current microphone input level.
I am not allowed to use external audio libraries, but I can use Windows libraries. So I tried using waveIn functions, but I do not know how to process audio input data in real time.
This is the method I am currently using:
Record for 100 milliseconds
Select highest value from the recorded data buffer
Repeat forever
But I think this is way too hacky, and not a recommended way. How can I do this properly?
Having built a tuning wizard for a very dated, but well known, A/V conferencing applicaiton, what you describe is nearly identical to what I did.
A few considerations:
Enqueue 5 to 10 of those 100ms buffers into the audio device via waveInAddBuffer. IIRC, when the waveIn queue goes empty, weird things happen. Then as the waveInProc callbacks occurs, search for the sample with the highest absolute value in the completed buffer as you describe. Then plot that onto your visualization. Requeue the completed buffers.
It might seem obvious to map the sample value as follows onto your visualization linearly.
For example, to plot a 16-bit sample
// convert sample magnitude from 0..32768 to 0..N
length = (sample * N) / 32768;
DrawLine(length);
But then when you speak into the microphone, that visualization won't seem as "active" or "vibrant".
But a better approach would be to give more strength to those lower energy samples. Easy way to do this is to replot along the μ-law curve (or use a table lookup).
length = (sample * N) / 32768;
length = log(1+length)/log(N);
length = max(length,N)
DrawLine(length);
You can tweak the above approach to whatever looks good.
Instead of computing the values yourself, you can rely on values from Windows. This is actually the values displayed in your screenshot from the Windows Settings.
See the following sample for the IAudioMeterInformation interface:
https://learn.microsoft.com/en-us/windows/win32/coreaudio/peak-meters.
It is made for the playback but you can use it for capture also.
Some remarks, if you open the IAudioMeterInformation for a microphone but no application opened a stream from this microphone, then the level will be 0.
It means that while you want to display your microphone peak meter, you will need to open a microphone stream, like you already did.
Also read the documentation about IAudioMeterInformation it may not be what you need as it is the peak value. It depends on what you want to do with it.

Measure difference between two files

I have a question that's very specific, yet very general at the same time. (Also, I don't know if this is quite the right site for this.)
The Scenario
Let's say I have an uncompressed video vid.avi. It is then run through [Some compression algorithm], which is lossy. I want to compare vid.avi and the new, compressed file to determine just how much data was lost in the compression. How can I compare the files and how can I measure the difference between the two, using the original as the reference point? Is it possible at all? I would prefer a generic answer that will work with any language, but I would also gladly accept an answer that's specific to a language.
EDIT: Let me be more specific. I want something that compares two video files in a similar way that the Notepad++ Compare plugin compares text files. I just want to find out how close each individual pixel's colour is to the original file's colour for that pixel.
Thanks in advance, and thank you for taking the time to read this question.
It is generally the change in video quality that people want to measure when comparing compression methods, rather than a loss of data.
If you did want to measure somehow the data loss, you would have to define what you mean by 'data' and how you wanted to measure it. Video compression is quite complex and the approach may even differ frame by frame within a video. Data could mean the colour depth for each pixel, the number of frames per second, whether a frame is encoded based on a delay to other frames etc.
Video quality is subjective so the reduction in quality after compression will not be an absolute value. The usual way to measure the quality is similar to the technique used for audio - Mean Opinion Score: https://en.wikipedia.org/wiki/Mean_opinion_score. Its essentially uses a well defined process to try to apply some objectivity to a test audiences subjective experience.

YIN-Frequency-Detection and overtones (guitar strings)

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

Mobile geolocation precision - cordova / phonegap

I want to develop an app that detects how far the user/device is from points on a map.
Calculating the distance is easy, but when you get close to about 30meters I would need it to be as precise as possible.
Basically I want some lights on the UI to get brighter the closer you get to the target/point.
How do I achieve this if the gps position sometimes bounces around for 5-10 meters or more?
Any ideas on how to approach this?
Thanks!
In general there is the inaccuracy with the position, and indeed its meters, thus the bouncing will be there and its rather impossible to get rid of it, anyways, one suggestion would be to collect the last few (3-10 up to you and your logic really) locations and calculate average from them. Then with fast movements your position would be lagging of course, but when doing slow movements the position shown should be more stable.. Of course you could also have additional logic on determining the movement direction, and accepting the location change towards that faster etc.
You will not get a better precision than 3m to the target.
At low, speed, like walking, you will no make it better than 8-10m.
Count the distance sicne last used fix, If it exceeds 12m then use the fix, and mark it as last used.
This is a simple filter which works well for walking speeds.
At speeds higher (> 10km/h) switch off the filter.
GPS should not jump at that speed.

OpenGL -- GL_LINE_LOOP --

I am using GL_LINE_LOOP to draw a circle in C and openGL! Is it possible for me to fill the circle with colors?
If needed, this is the code I'm using:
const int circle_points=100;
const float cx=50+i, cy=50+x, r=50;
const float pi = 3.14159f;
int i = 50;
glColor3f(1, 1, 1);
glBegin(GL_LINE_LOOP);
for(i=0;i<circle_points;i++)
{
const float theta=(2*pi*i)/circle_points;
glVertex2f(cx+r*cos(theta),cy+r*sin(theta));
}
glEnd();
Lookup polygon triangulation!
I hope something here is somehow useful to someone, even though this question was asked in February. There are many answers, even though a lot of people would give none. I could witter forever, but I'll try to finish before then.
Some would even say, "You never would," or, "That's not appropriate for OpenGL," I'd like to say more than them about why. Converting polygons into the triangles that OpenGL likes so much is outside of OpenGL's job-spec, and is probably better done on the processor side anyway. Calculate that stage in advance, as few times as possible, rather than repeatedly sending such a chunky problem on every draw call.
Perhaps the original questioner drifted away from OpenGL since February, or perhaps they've become an expert. Perhaps I'll re-inspire them to look at it again, to hack away at some original 'imposters'. Or maybe they'll say it's not the tool for them after all, but that would be disappointing. Whatever graphics code you're writing, you know that OpenGL can speed it up!
Triangles for convex polygons are easy
Do you just want a circle? Make a triangle fan with the shared point at the circle's origin. GL_POLYGON was, for better or worse, deprecated then killed off entirely; it will not work with current or future implementations of OpenGL.
Triangles for concave polygons are hard
You'll want more general polygons later? Well, there are some tricks you could play with, for all manner of convex polygons, but concave ones will soon get difficult. It would be easy to start five different solutions without finishing a single one. Then it would be difficult, on finishing one, to make it quick, and nearly impossible to be sure that it's the quickest.
To achieve it in a future-proofed way you really want to stick with triangles -- so "polygon triangulation" is the subject you want to search for. OpenGL will always be great for drawing triangles. Triangle strips are popular because they reuse many vertices, and a whole mesh can be covered with only triangle strips, (perhaps including the odd lone triangle or pair of triangles). Drawing with only one primitive usually means the entire mesh can be rendered with a single draw call, which could improve performance. (Number of draw calls is one performance consideration, but not always the most important.)
Polygon triangulation gets more complex when you allow convex polygons or polygons with holes. (Finding algorithms for triangulating a general polygon, robustly yet quickly, is actually an area of ongoing research. Nonetheless, you can find some pretty good solutions out there that are probably fit for purpose.)
But is this what you want?
Is a filled polygon crucial to your final goals in OpenGL? Or did you simply choose what felt like it would be a simple early lesson?
Frustratingly, although drawing a filled polygon seems like a simple thing to do -- and indeed is one of the simplest things to do in many languages -- the solution in OpenGL is likely to be quite complicated. Of course, it can be done if we're clever enough -- but that could be a lot of effort, without being the best route to take towards your later goals.
Even in languages that implement filled polygons in a way that is simple to program with, you don't always know how much strain it puts on the CPU or GPU. If you send a sequence of vertices, to be linked and filled, once every animation frame, will it be slow? If a polygon doesn't change shape, perhaps you should do the difficult part of the calculation just once? You will be doing just that, if you triangulate a polygon once using the CPU, then repeatedly send those triangles to OpenGL for rendering.
OpenGL is very, very good at doing certain things, very quickly, taking advantage of hardware acceleration. It is worth appreciating what it is and is not optimal for, to decide your best route towards your final goals with OpenGL.
If you're looking for a simple early lesson, rotating brightly coloured tetrahedrons is ideal, and happens early in most tutorials.
If on the other hand, you're planning a project that you currently envision using filled polygons a great deal -- say, a stylized cartoon rendering engine for instance -- I still advise going to the tutorials, and even more so! Find a good one; stick with it to the end; you can then think better about OpenGL functions that are and aren't available to you. What can you take advantage of? What do you need or want to redo in software? And is it worth writing your own code for apparently simple things -- like drawing filled polygons -- that are 'missing from' (or at least inappropriate to) OpenGL?
Is there a higher level graphics library, free to use -- perhaps relying on OpenGL for rasterisation -- that can already do want you want? If so, how much freedom does it give you, to mess with the nuts and bolts of OpenGL itself?
OpenGL is very good at drawing points, lines, and triangles, and hardware accelerating certain common operations such as clipping, face culling, perspective divides, perspective texture accesses (very useful for lighting) and so on. It offers you a chance to write special programs called shaders, which operate at various stages of the rendering pipeline, maximising your chance to insert your own unique cleverness while still taking advantage of hardware acceleration.
A good tutorial is one that explains the rendering pipeline and puts you in a much better position to assess what the tool of OpenGL is best used for.
Here is one such tutorial that I found recently: Learning Modern 3D Graphics Programming
by Jason L. McKesson. It doesn't appear to be complete, but if you get far enough for that to annoy you, you'll be well placed to search for the rest.
Using imposters to fill polygons
Everything in computer graphics is an imposter, but the term often has a specialised meaning. Imposters display very different geometry from what they actually have -- only more so than usual! Of course, a 3D world is very different from the pixels representing it, but with imposters, the deception goes deeper than usual.
For instance, a rectangle that OpenGL actually constructs out of two triangles can appear to be a sphere if, in its fragment shader, you write a customised depth value to the depth coordinate, calculate your own normals for lighting and so on, and discard those fragments of the square that would fall outside the outline of the sphere. (Calculating the depth on those fragments would involve a square root of a negative number, which can be used to discard the fragment.) Imposters like that are sometimes called flat cards or billboards.
(The tutorial above includes a chapter on imposters, and examples doing just what I've described here. In fact, the rectangle itself is constructed only part way through the pipeline, from a single point. I warn that the scaling of their rectangle, to account for the way that perspective distorts a sphere into an ellipse in a wide FOV, is a non-robust fudge . The correct and robust answer is tricky to work out, using mathematics that would be slightly beyond the scope of the book. I'd say it is beyond the author's algebra skills to work it out but I could be wrong; he'd certainly understand a worked example. However, when you have the correct solution, it is computationally inexpensive; it involves only linear operations plus two square roots, to find the four limits of a horizontally- or vertically-translated sphere. To generalise that technique for other displacements requires one more square root, for a vector normalisation to find the correct rotation, and one application of that rotation matrix when you render the rectangle.)
So just to suggest an original solution that others aren't likely to provide, you could use an inequality (like x * x + y * y <= 1 for a circle or x * x - y * y <= 1 for a hyperbola) or a system of inequalities (like three straight line forms to bound a triangle) to decide how to discard a fragment. Note that if inequalities have more than linear order, they can encode perfect curves, and render them just as smoothly as your pixelated screen will allow -- with no limitation on the 'geometric detail' of the curve. You can also combine straight and curved edges in a single polygon, in this way.
For instance, a fragment shader (which would be written in GLSL) for a semi-circle might have something like this:
if (y < 0) discard;
float rSq = x * x + y * y;
if (1 < rSq) discard;
// We're inside the semi-circle; put further shader computations here
However, the polygons that are easy to draw, in this way, are very different from the ones that you're used to being easy. Converting a sequence of connected nodes into inequalities means yet more code to write, and deciding on the Boolean logic, to deal with combining those inequalities, could then get quite complex -- especially for concave polygons. Performing inequalities in a sensible order, so that some can be culled based on the results of others, is another ill-posed headache of a problem, if it needs to be general, even though it is easy to hard-code an optimal solution for a single case like a square.
I suggest using imposters mainly for its contrast with the triangulation method. Something like either one could be a route to pursue, depending on what you're hoping to achieve in the end, and the nature of your polygons.
Have fun...
P.S. have a related topic... Polygon triangulation into triangle strips for OpenGL ES
As long as the link lasts, it's a more detailed explanation of 'polygon triangulation' than mine. Those are the two words to search for if the link ever dies.
A line loop is just an outline.
To fill the middle as well, you want to use GL_POLYGON.

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