Hello I was wondering if using the android tone generator class would it be possible to create a tone in one device and listen for this same tone in another device. If this is possible I do have a few other questions.
Taking backround noise into consideration is it possible to listen for only this specific tone?
Would this process be resource intensive?
Could I use a tone that would be inaudable to the human ear or close to it?
Lastly could I use a tone that could only be heard with a couple of feet from the sending device?
Thanks very much for yer time guys and girls :)
Edit >
Thanks For adding the audio processing tag sabastian. Much better discription.
It would be CPU intensive, yes.
The way to it is quite simple: you need a permanent recorder which puts the received data into a FFT (fast fourier transform). FFT basically does one thing: splits the audio into a frequency/power-scale. With this "background noise cleaned" result you can check things like "was there a tone with 1000Hz playing for at least 2 seconds" - and act accordingly.
There is a reasonable speed FFT implementation here: http://www.badlogicgames.com/wordpress/?p=449
FFT can also be used (actually, IS used) for detection of dualtone dialing (DTMF) - 2 frequencies at same time is much better than just using one (as the error rate drop significantly and you can go to shorter duration for the tone sending/detecting).
"Inaudible" won't be possible, as (a) the speaker can not produce such sounds (b) you are limited in sampling rate - so also limited in both producing and recording such high frequencies.
"couple of feet" will be naturally imposed (not very loud speaker, not very good microphone).
Have a look at this other question: "Android: Need to record mic input". I think you can modify that for your task, then with sound bytes you can have filtering or FFT.
Hope it helps
Related
Is it possible to make flash blink on ringtone, according to the beats of a song set as ringtone on phone in android.If yes then please give me a hint.
Yes and no. I've once written a FFT for acceleration sensor data of the devices. This was to find out what step frequence a user is running.
You can do FFT on an Android phone. I used the Apache Commons Math library for it.
Easiest starting point is to have WAV or FLAC Audio files as they do not have compressions or the like in it. You can straight do the FFT on them.
In my running app it takes about 1-2 seconds(!) to calculate one FFT for 512 input values, utilizing the CPU for 100% and draining the battery correspondingly. So I do the FFT only once a minute.
I can assume, that the user would see this, when the telephone rings:
No blinking light, just sound for the first view seconds (while you do the FFT)
The blinking light with the frequence of the beats but not in sync with them (you're showing the beats of a past time period)
But I must admit, that I really like the idea. Let me know, when you're out with your App :-)
I really fail at FFT and now I'm in need to communicate from the headphone jack of my Android to the Arduino there's currently a library for Arduino (talks about it in the blog post Real-time spectrum analyzer powered by Arduino) and one for Android too!
How should I start? How should I build audio signals which ultimately can be turned into FFTs and the Arduino can analyse the same using the library and I can actuate anything?
You are asking a very fuzzy question: "How should I build audio signals which ultimately can be turned into FFTs and the Arduino can analyse the same using the library and I can actuate anything?". I am going to help you think through the problem - asking yourself the right questions is essential to get any answers.
Presumably, your audio signals are "coming from somewhere" - i.e. they are sound. This means that you need to convert them into a stream of numbers first.
problem #1: converting audio signal into a stream of numbers
This breaks down into three separate sub problems:
Getting the signal to the right amplitude
Choosing the sampling rate needed
Digitizing and storing the data for later processing
Items (1) and (3) are related, since you need to know how you are going to digitize the signal before you can choose the right amplitude. For example, if you have a microphone as your sound input source, you will need to amplify the signal (and maybe add some automatic gain control) before feeding it into an ADC (analog to digital converter) that has a 5 V input range, since the microphone may have an output in the mV range. Without more information about the hardware you are using, there's not a lot to add here. It sounds from your tag that you are trying to do that inside an Android device - in which case I wonder how you intend to move the digital signal to the Arduino (over USB?).
The second point, "choosing the sampling rate", is actually very important. A sound signal contains many different frequencies - think of them as keys on the piano. In order to detect a high frequency, you need to sample the signal "faster than it is changing". There is a formal theorem called "Nyquist's Theorem" that states that you have to sample at 2x the highest frequency that is present in your signal. Note - it's not just "that you are interested in", but "that is present". If you sample a high frequency signal with a low frequency sample clock, it will appear "aliased" - it wil show up in your output as something completely different. So before you digitize a signal you have to decide what the frequencies of interest are, and remove all higher frequencies with a filter. Let's say you are interested in frequencies up to 500 Hz (about 1 octave above middle C on a piano). To give your filter a chance to work, you might choose to cut off all frequencies above 1 kHz (filters "roll off" - i.e. they increase in strength over a range of frequencies), and would sample at 2 kHz. This means you get 2000 samples per second, and you need to figure out where to put them on your Arduino (memory fills up quickly on the little board.)
Problem #2: analyzing the signal
Assuming that you have somehow captured a digital signal, your next task is analyzing it. The FFT is basicaly some clever math that tells you, for a given sound sample, "what keys on the piano were hit, and how hard". It breaks the sound signal into a series of frequency "bins", and determines how much energy is in each bin (it also computes the phase, but let's keep it simple). So if the input of a FFT algorithm is a sound sample, the output is an array of values telling you what frequencies were present in the signal. This is approximate, since it will find the "nearest bin". Sticking with the same analogy - if you were hitting a piano that's out of tune, the algorithm won't return "out of tune", but rather "a bit of C, and a bit of C sharp", since it cannot actually measure anything in between. The accuracy of an FFT is determined by the sampling frequency (which gives you the upper limit on the frequency you can detect) and the sample length: the longer you "listen" so the sample, the more subtle the differences you can "hear". So you have another trade-off to consider: if your audio signal changes rapidly, you have to sample for a short time (to capture the quick changes); but if you need an accurate frequency, you have to sample for a long time. For example if you are writing a Morse decoder, your sampling has to be short compared to a pause between "dits" and "dashes" - or they will slur together. Figuring out that a morse tone is present is pretty easy though, since there will be a single tone (one bin in the FFT) that is much larger than the others.
Exactly how you implement these things depends on your application. The third step, "doing something with it", requires you to decide what is a meaningful signal. Again, if you are making a Morse decoder, you would perhaps turn an LED ON when a single tone is present (one or two bins in the FFT have much bigger value than the mean of the others), and OFF when it is not (all noise - lots of bins with approximately the same size). But without a LOT more information from you, there's not much more one can say to help you.
You might learn a lot from reading the following articles:
http://www.arduinoos.com/2010/10/sound-capture/
http://www.arduinoos.com/2010/10/fast-fourier-transform-fft/
http://interface.khm.de/index.php/lab/experiments/frequency-measurement-library/
How can I detect when the user blows into the device microphone? This would then be used to trigger some action by the app.
The job of detecting when a user blows into the microphone is separable into two parts: (1) taking input from the microphone and (2) listening for a blowing sound.
The noise/sound of someone blowing into the mic is made up of low-frequency sounds. We’ll use a low pass filter to reduce the high frequency sounds coming in on the mic; when the level of the filtered signal spikes we’ll know someone’s blowing into the mic.
Source:
http://mobileorchard.com/tutorial-detecting-when-a-user-blows-into-the-mic/
EDIT
And here is some small SoundMeter class for Android:
http://code.google.com/p/android-labs/source/browse/trunk/NoiseAlert/src/com/google/android/noisealert/SoundMeter.java?r=2
I would make and FFT and compare the spectrum with that of more sensible sounds. The blow will likely resemble white noise. Before seeing the spectrums of blow, speech and white noise, I have no idea how to tell one from another.
I am writing an application that will behave similar to the existing Voice recognition but will be sending the sound data to a proprietary web service to perform the speech recognition part. I am using the standard MediaRecord (which is AMR-NB encoded) which seems to be perfect to speech recognition. The only data provided by this is the Amplitude via the getMaxAmplitude() method.
I am trying to detect when the person starts to talk so that when the person stops talking for about 2 seconds I can proceed to send the sound data to the web service. Right now I am using a threshold for the amplitude that if its goes over a value (i.e. 1500) then I assume the person is speaking. My concern is that the amplitude levels may vary by device (i.e. Nexus One v Droid), so I am looking for a more standard approach to this that can be derived from the amplitude values.
P.S.
I looked at graphing-amplitude but it doesn't provide a way to do it with just the amplitude.
Well, this might not be of much help but how about starting by measuring the offset noise captured by the microphone of the device by the application, and apply the threshold dynamically based on that? That way you would make it adaptable to the different devices' microphones and also to the environment the user is using it at, at a given time.
1500 is too low of a number. Measuring the change in amplitude will work better.
However, it will still result in miss detections.
I fear the only way to solve this problem is to figure out how to recognize a simple word or tone rather than simply detect noise.
There are now multiple VAD library designed for Android. One of these are:
https://github.com/gkonovalov/android-vad
Most of the smartphones come with a proximity sensor. Android has API for using these sensors. This would be adequate for the job you described. When the user moves the phone near to his ear, you can code the app to start recording. It should be easy enough.
Sensor class for android
I'm trying to build a gadget that detects pistol shots using Android. It's a part of a training aid for pistol shooters that tells how the shots are distributed in time and I use a HTC Tattoo for testing.
I use the MediaRecorder and its getMaxAmplitude method to get the highest amplitude during the last 1/100 s but it does not work as expected; speech gives me values from getMaxAmplitude in the range from 0 to about 25000 while the pistol shots (or shouting!) only reaches about 15000. With a sampling frequency of 8kHz there should be some samples with considerably high level.
Anyone who knows how these things work? Are there filters that are applied before registering the max amplitude. If so, is it hardware or software?
Thanks,
/George
It seems there's an AGC (Automatic Gain Control) filter in place. You should also be able to identify the shot by its frequency characteristics. I would expect it to show up across most of the audible spectrum, but get a spectrum analyzer (there are a few on the app market, like SpectralView) and try identifying the event by its frequency "signature" and amplitude. If you clap your hands what do you get for max amplitude? You could also try covering the phone with something to muffle the sound like a few layers of cloth
It seems like AGC is in the media recorder. When I use AudioRecord I can detect shots using the amplitude even though it sometimes reacts on sounds other than shots. This is not a problem since the shooter usually doesn't make any other noise while shooting.
But I will do some FFT too to get it perfect :-)
Sounds like you figured out your agc problem. One further suggestion: I'm not sure the FFT is the right tool for the job. You might have better detection and lower CPU use with a sliding power estimator.
e.g.
signal => square => moving average => peak detection
All of the above can be implemented very efficiently using fixed point math, which fits well with mobile android platforms.
You can find more info by searching for "Parseval's Theorem" and "CIC filter" (cascaded integrator comb)
Sorry for the late response; I didn't see this question until I started searching for a different problem...
I have started an application to do what I think you're attempting. It's an audio-based lap timer (button to start/stop recording, and loud audio noises for lap setting). It' not finished, but might provide you with a decent base to get started.
Right now, it allows you to monitor the signal volume coming from the mic, and set the ambient noise amount. It's also using the new BSD license, so feel free to check out the code here: http://code.google.com/p/audio-timer/. It's set up to use the 1.5 API to include as many devices as possible.
It's not finished, in that it has two main issues:
The audio capture doesn't currently work for emulated devices because of the unsupported frequency requested
The timer functionality doesn't work yet - was focusing on getting the audio capture first.
I'm looking into the frequency support, but Android doesn't seem to have a way to find out which frequencies are supported without trial and error per-device.
I also have on my local dev machine some extra code to create a layout for the listview items to display "lap" information. Got sidetracked by the frequency problem though. But since the display and audio capture are pretty much done, using the system time to fill in the display values for timing information should be relatively straightforward, and then it shouldn't be too difficult to add the ability to export the data table to a CSV on the SD card.
Let me know if you want to join this project, or if you have any questions.