audio - Android AudioRecord filter range of frequency -

i using android platform, following reference question come know using audiorecord class returns raw data can filter range of audio frequency depends upon need need algorithm, can please me out find algorithm filter range b/w 14,400 bph , 16,200 bph.

i tried "jtransform" don't know can achieve jtransform or not ? using "jfftpack" display visual effects works can't achieve audio filter using this.

reference here

help appreciated in advance. following code mentioned above using "jfftpack" library display may find library reference in code please don't confuse that

private class recordaudio extends asynctask<void, double[], void> {         @override         protected void doinbackground(void... params) { try {     final audiorecord audiorecord = findaudiorecord();                     if(audiorecord == null){                         return null;                     }                      final short[] buffer = new short[blocksize];                     final double[] totransform = new double[blocksize];                      audiorecord.startrecording();       while (started) {                         final int bufferreadresult = audiorecord.read(buffer, 0, blocksize);                          (int = 0; < blocksize && < bufferreadresult; i++) {                             totransform[i] = (double) buffer[i] / 32768.0; // signed 16 bit                         }                          transformer.ft(totransform);                         publishprogress(totransform);                      } audiorecord.stop();                 audiorecord.release(); } catch (throwable t) {                 log.e("audiorecord", "recording failed");             }             return null;  /**          * @param totransform          */         protected void onprogressupdate(double[]... totransform) {             canvas.drawcolor(color.black);             (int = 0; < totransform[0].length; i++) {                 int x = i;                 int downy = (int) (100 - (totransform[0][i] * 10));                 int upy = 100;                 canvas.drawline(x, downy, x, upy, paint);             }             imageview.invalidate();         } 

there lot of tiny details in process can potentially hang here. code isn't tested , don't audio filtering should extremely suspicious here. basic process take filtering audio:

1. get audio buffer
2. possible audio buffer conversion (byte float)
3. (optional) apply windowing function i.e. hanning
4. take fft
5. filter frequencies
6. take inverse fft

i'm assuming have basic knowledge of android , audio recording cover steps 4-6 here.

//it assumed float array audiobuffer exists length =  //the capture size of audio buffer  //the size of fft size of audiobuffer / 2 int fft_size = buffersize / 2; floatfft_1d mfft = new floatfft_1d(fft_size); //this jtransforms type  //take fft mfft.realforward(audiobuffer);  //the first 1/2 of audiobuffer contains bins represent frequency //of wave, in way.  actual frequency bin: //frequency_of_bin = bin_index * sample_rate / fft_size  //assuming length of audiobuffer even, real , imaginary parts //stored follows //audiobuffer[2*k] = re[k], 0<=k<n/2 //audiobuffer[2*k+1] = im[k], 0<k<n/2  //define frequencies of interest float freqmin = 14400; float freqmax = 16200;  //loop through fft bins , filter frequencies for(int fftbin = 0; fftbin < fft_size; fftbin++){             //calculate frequency of bin assuming sampling rate of 44,100 hz     float frequency = (float)fftbin * 44100f / (float)fft_size;      //now filter audio, i'm assuming wanted keep     //frequencies of interest rather discard them.     if(frequency  < freqmin || frequency > freqmax){         //calculate index real , imaginary parts stored         int real = 2 * fftbin;         int imaginary = 2 * fftbin + 1;          //zero out frequency         audiobuffer[real] = 0;         audiobuffer[imaginary] = 0;     } }  //take inverse fft convert signal frequency time domain mfft.realinverse(audiobuffer, false);