I want to find out the level of noise around the phone. There doesn't seem to be an easy built-in way to do this, so I've found a few examples that say to use AudioRecord to listen for a brief time and then to use a formula to get the decibel. I have a few questions that the documentation doesn't seem much to explaining though and I was wondering if you kind folk could help me understand.
What's captured in the Array? AudioRecord takes an several arguments on Channel type, format and sample rate and stores that information into an array. I can see the raw numbers, but what do they actually mean?
Is there a resource that your could point me to, or explain to me, on how to convert(what I assume to be) the raw audio in byte form into a decibel representation?
The part I don't understand is #1, what is actually put in the array. Any help would be appreciated.
Found a great resource that converts to a 0-9 range the amplitude of the sound at the mic.
SoundMeter is done by google and works well enough for my purpose.
Related
I am writing an android app in which I want to convert the signal amplitude coming from microphone to its corresponding rms voltage.
I know we can use getMaxAmplitude(). But I don't know how to convert the value that we get from getMaxAmplitude() to a RMS voltage.
I researched a lot but didn't find anything. Thanks.
Generally the device will supply voltage to the microphone.
First, to answer your question: I believe you won't find a way to get the magnitude of this voltage from the Android API. If you will find anything it will beon the NDK side of things, deep in the driver somewhere, hence the each device will be different.
I will presume you're using a device that expects an electret microphone as I am sure you know, this is how it generally works:
Each device will supply slightly different voltage, so you will probably have to measure it. I have seen 2.5V, 3.2V or 4V, sometimes those values are adjustable from the softwareside in drivers, but again each device will work differently.
These things are not really designed to be run as scopes, but it is doable, especially if you only target one specific device.
Sorry for the bad news and good luck with your project ;)
I'm using Android's MediaCodec class to read raw data from audio files. That works just fine.
The problem is that I don't know if it's safe to assume that the output data will always be 16-bit?
I can tell, experimentally, that the output is 16-bit, but I don't know how to check that at runtime. The MediaCodec documentation doesn't appear to tell me. The MediaFormat KEY_CHANNEL_MASK could tell me, but MediaCodec doesn't appear to set those flags. It sets the sample rate, and the mime-type, but nothing that can tell me the bit-size explicitly.
I suppose that given the difference between presentation times of subsequent blocks, and the sample rate, I should be able to calculate it, but that doesn't seem very satisfactory.
Is there a way to tell, or is it written somewhere that I don't have to?
Currently the output is always 16 bit in stock Android. If that changes in the future we'll add an additional format key that specifies the format. Note that KEY_CHANNEL_MASK would only tell you which channels are included (e.g. left, right, center, etc), not the sample format.
No it does not. If you have 24 bit PCM (WAV) files, it will give you 24 bit audio. And seemingly no way to determine this. I have added a question here MediaCodec and 24 bit PCM
In my android application I need to capture the user's speech from the microphone and then pass it to the server. Currently, I use the MediaRecorder class. However, it doesn't satisfy my needs, because I want to make glowing effect, based on the current volume of input sound, so I need an AudioStream, or something like that, I guess. Currently, I use the following:
this.recorder = new MediaRecorder();
this.recorder.setAudioSource(MediaRecorder.AudioSource.MIC);
this.recorder.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4);
this.recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB);
this.recorder.setOutputFile(FILENAME);
I am writing using API level 7, so I don't see any other AudioEncoders, but AMR Narrow Band. Maybe that's the reason of awful noise which I hear in my recordings.
The second problem I am facing is poor sound quality, noise, so I want to reduct (cancel, suppress) it, because it is really awful, especially on my noname chinese tablet. This should be server-side, because, as far as I know, requiers a lot of resources, and not all of the modern gadgets (especially noname chinese tablets) can do that as fast as possible. I am free to choose, which platform to use on the server, so it can be ASP.NET, PHP, JSP, or whatever helps me to make the sound better. Speaking about ASP.NET, I have come across a library, called NAudio, may be it can help me in some way. I know, that there is no any noise reduction solution built in the library, but I have found some examples on FFT and auto-corellation using it, so it may help.
To be honest, I have never worked with sound this close before and I have no idea where to start. I have googled a lot about noise reduction techniques, code examples and found nothing. You guys are my last hope.
Thanks in advance.
Have a look at this article.
Long story short, it uses MediaRecorder.AudioSource.VOICE_RECOGNITION instead of AudioSource.MIC, which gave me really good results and noise in the background did reduce very much.
The great thing about this solution is, it can be used with both AudioRecord and MediaRecorder class.
For audio capture you can use the AudioRecord class. This lets you record raw audio, i.e. you are not restricted to "narrow band" and you can also measure the volume.
Many smartphones have two microphones, one is the MIC you are using, the other one is near camera for video shooting, called CAMCORDER. You can get data from both of them to do noise reduction. There are many papers talking about audio noise reduction with multiple microphones.
Ref: http://developer.android.com/reference/android/media/MediaRecorder.AudioSource.html
https://www.google.com/search?q=noise+reduction+algorithm+with+two+mic
I made test application in Delphi that beeps morse code using Windows API Beep function. Then made an application in Android that stores this morse code in WAV file. Now I want Android application to decode the morse code. Is there some tutorials for sound processing or can somebody post some simple code (think there's no simplicity here) for an example? Or maybe steps that I need to do to get it work?
I also downloaded the JTransforms and jfttw libraries but don't really know where to start.
Regards,
evilone
An FFT is overkill for this - you can just use a simple Goertzel filter to isolate the morse code from background noise, then decode the output of this.
I think an older issues of QST magazine had an article on DSP for Morse/CW decoding several years back. Might want to try and search their archives.
Basically, you need DSP code to determine whether or not a tone is present at any given point in time, and an estimate of the onset and off-time of each tone. Then scale the duration of each tone and the gap times between the tones for the expected code speed, and compare against a table of timings for each Morse code letter to estimate the probability of each or any letter being present.
In the simplest case, you might have a dot-dash-space decision tree. In severe noise and fading plus highly personalized fist/timing you might need some sophisticated statistical and/or adaptive audio pattern matching techniques for decent results.
I have searched for this online, but am still a bit confused (as I'm sure others will be if they think of something like this). I'd like to preface by saying that this is not for homework and/or profit.
I wanted to create an app that could listen to your microwave as you prepare popcorn. It would work by sounding an alarm when there's a certain time interval between pops (say 5-6 seconds). Again, this is simply a project to keep me occupied - not for a class.
Either way, I'm having trouble trying to figure out how to analyze the audio intake in real-time. That is, I need a way to log the time when a "pop" occurs. So that you guys don't think I didn't do any research into the matter, I've checked out this SO question and have extensively searched the AudioRecord function list.
I'm thinking that I will probably have to do something with one of the versions of read() and then compare the recorded audio every 2 seconds or so to the recorded audio of a "pop" (i.e. if 70% or more of the byte[] audioData array is the same as that of a popping sound, then log the time). Can anyone with Android audio input experience let me know if I'm at least on the right track? This is not a question of me wanting you to code anything for me, but a question as to whether I'm on the correct track, and, if not, which direction I should head instead.
I think I have an easier way.
You could use the MediaRecorder 's getMaxAmplitude method.
Anytime your recorder detects a big jump in amplitude, you have detected a corn pop!
Check out this code (ignore the playback part): Playing back sound coming from microphone in real-time
Basically the idea is that you will have to take the value of each 16-bit sample (which corresponds to the value of the wave at that time). Using the sampling rate, you can calculate the time between peaks in volume. I think that might accomplish what you want.
this may be a bit overkill, but there is a framework from MIT media labs called funf: http://code.google.com/p/funf-open-sensing-framework/
They already created classes for audio input and some analysis (FFT and the like), also saving to files or uploading is implemented as far as I've seen, and they handle most of the sensors available on the phone.
You can also get inspired from the code they wrote, which I think is pretty good.