For an app I'm working on a i need to be able to 'search' a sound file to find a particular frequency.
Basically, the iPhone mic records for 5 seconds writes to an a lossless music file. I then need to 'open' that file and search for a particular frequency. The frequency is very particular (eg not between 15hz and 300hz, its is a fixed number).
Also, as soon the the frequency is found the search can stop.
I've got the iPhone recording the sound and writing to the file, I am just unsure how to open that file and search for the frequency.
Any help would be greatly appreciated!
Thanks
Checkout PitchDetector and the related tutorial.
Can't get you any closer :)
If you're just looking for a specific, fixed, pure tone then the simplest method is probably Goertzel's algorithm - it's very simple to implement and is relatively lightweight computationally compared to using e.g. an FFT or autocorrelation method.
Related
I am working on making an app that performs an action when the sound of a clap is recognized. I have looked into simply measuring the average and peak power from an AVAudioRecorder and this works okay, but if there are other sounds then it reports lots of false positives. I believe I need some kind of audio fingerprinting for this to work while other audio is playing. Now I know that this has been asked a lot before on SO, but most of the answers say something along the lines of "Use FFT" and then the person says "Oh okay!" but no clear explanation is given and I still have no idea how to correctly identify sounds using an FFT.
Can anyone clearly explain, cite another tutorial, or post a link to a library that can identify sounds using audio fingerprinting?
Thanks!
After finally successfully finding a way to concatenate multiple voice files into one single audio file on the iPhone, I am am now trying to superimpose an audio file over the length of the voice file.
So basically I have two .m4a files:
voice.m4a which is about 10 seconds for example.
music.m4a which is about 5 seconds.
What I require is that two file be combined in such a manner that the resulting single audio file now contains the music in the background of the voice file for the length of it, so basically the resulting output should have the 10 seconds of voice and the 5seconds of music repeated twice. It is absolutely important to have a single file that contains all of this.
I am trying to get all of this done in an application on the iPhone.
Can anyone please help me out with this?
If you are looking to do that programmatically, you will need to go deeper down into CoreAudio. For a simpler solution you could use AudioQueues or for more fine grained control AudioUnits and an AUGraph. The MultiChannelMixer is the Audio Unit you are looking for. Unfortunately there is no space for an elaborate tutorial here (would take a couple of days to write just the tutorial itself), but I am hoping I could point you to the right direction.
If you decide to go down that path and want to do further audio programming then this one time simple example, then I strongly suggest you buy "Learning Core Audio, A Hands-on Guide to Audio Programming for Mac and iOS" - Chris Adamson, Kevin Avila. You can find it on Amazon, paperback or Kindle.
guys.
I'm working on some audio services on iOS.
I trying to search any examples or tutorials about
how audio service or stream can read a existing audio file than
process something like filter, than write another file.
Is there any body who can help me?
Dirac3LE (by Stephan M. Bernsee) is a great library for this job.
There are examples and manual included in the download.
It is particulary inteded for time and pitch manipulation
but in your case you'll be interested in its EAFRead and EAFWrite
classes.
If you want to get familiar with the lower level library that you can also use for microphone input/sound output, and that you can get raw samples into and out of, I would suggest taking a look at Audio Queue Services.
I used it in my side project to get audio from the microphone, and I also wrote some code you might find useful to do fast vectorized, FFT based FIR filtering on input audio. You can find the code here https://github.com/jamescarlson/FreeAPRS
I've been searching for some examples that show how to do ADSR in iOS using audio samples (preferably WAV files with loop points, but thats secondary). I guess most people who write a sampler/synth app use audio unit for this. Does any one know a good code example that shows ADSR in any iOS audio library?
In the new iOS SDK 5.0 there's now a Sampler Audio Unit! Which can do ADSR envelopes.
The presets demo shows how to use the sampler:
http://developer.apple.com/library/ios/#samplecode/LoadPresetDemo/Introduction/Intro.html#//apple_ref/doc/uid/DTS40011214
If you want to load different sound formats to play this article is helpful:
https://developer.apple.com/library/mac/#technotes/tn2283/_index.html
And here's the iOS documentation reference:
http://developer.apple.com/library/ios/#documentation/AudioUnit/Reference/AUComponentServicesReference/Reference/reference.html#//apple_ref/doc/uid/TP40007291
you can find (a very basic) one in the Apple's SinSynth sample. That is an AU, but it should demonstrate how one would apply a envelope to an audio buffer. i don't remember - it may simply be an ASR, but adding a fourth stage is simple once you have understood the existing program. The implementation is right in the note's render.
Envelope Generators are not platform specific.
musicdsp.org will be a better resource if you want more than a push in the right direction.
MusicDSP has source code for an example envelope follower with attack/release. If you understand this, then sustain/decay should be pretty logical. ;)
But an ADSR envelope is basically just a matter of applying gain to your output signal with a state machine. Each state has a starting value, and ending value, and a duration. Calculating the slope of that line and the value of each point along it was covered in your algebra class back in high school. ;) If you want to be really fancy, you can implement other types of curves, but the concept remains the same.
My aim is code a project which records human sound and changes it (with effects).
e.g : a person will record its sound over microphone (speak for a while) and than the program makes its like a baby sound.
This shall run effectively and fast (while recording the altering operation must run, too)
What is the optimum way to do it ?
Thanks
If you're looking for either XNA or DirectX to do this for you, I'm pretty sure you're going to be out of luck (I don't have much experience with DirectSound; maybe somebody can correct me). What it sounds like you want to do is realtime digital signal processing, which means that you're either going to need to write your own code to manipulate the raw waveform, or find somebody else who's already written the code for you.
If you don't have experience writing this sort of thing, it's probably best to use somebody else's signal processing library, because this sort of thing can quickly get complicated. Since you're developing for the PC, you're in luck; you can use any library you like using P/Invoke. You might try out some of the solutions suggested here and here.
MSDN has some info about the Audio namespace from XNA, and the audio recording introduced in version 4:
Working with Microphones
Recording Audio from a Microphone
Keep in mind that recorded data is returned in PCM format.