Question

Android: I am new to voice process, I have created AudioRecord Object with sampleRate 8000Hz using JTransform library i am trying to filter frequency there are couple of things in the following code which i do not understand my questions as follows

Q.1 Why we are converting “toTransform[i] = (double) buffer[i] / 32768.0; // signed 16 bit” buffer into 16bit value ?

Q.2 Rite now audioRecord read data array is short array if i read byte array how will i convert it into singed 16 bit ?

Q.3 I want to display sound frequency in Hz with double array how can i calculate sound frequency ?

Q.4 I wrote filter method filterAudio() but it is not filtering range of frequency ?

Please help me out i have lot of questions in my mind

/* Code as follows */

private final int[] mSampleRates = new int[] { 8000, 11025, 22050, 44100 };
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) {
                    Thread.sleep(100);
                    final int bufferReadResult = audioRecord.read(buffer, 0, blockSize);

                    for (int i = 0; i < blockSize && i < bufferReadResult; i++) {
                        toTransform[i] = (double) buffer[i] / 32768.0; // signed 16 bit
                    }

                    //Audio Filter passing frequency of mSampleRates[3]
                    filterAudio(bufferReadResult, toTransform, mSampleRates[3]);
                    transformer.realForward(toTransform);                   
                    publishProgress(toTransform);
                }
                audioRecord.stop();
                audioRecord.release();


public static void filterAudio(int bufferSize, double[] audioBuffer, float sampleRate ){

        //it is assumed that a float array audioBuffer exists with even length = to 
        //the capture size of your audio buffer

        //float frequency=0F;
        //The size of the FFT will be the size of your audioBuffer / 2
        int FFT_SIZE = bufferSize / 2;
        //RealDoubleFFT mFFT = new RealDoubleFFT(FFT_SIZE);
        DoubleFFT_1D mFFT = new DoubleFFT_1D(FFT_SIZE); //this is a jTransforms type

        //Take the FFT
        mFFT.realForward(audioBuffer);
        //mFFT.ft(audioBuffer);

        //The first 1/2 of audioBuffer now contains bins that represent the frequency
        //of your wave, in a way.  To get the actual frequency from the bin:
        //frequency_of_bin = bin_index * sample_rate / FFT_SIZE

        //assuming the length of audioBuffer is even, the real and imaginary parts will be
        //stored as follows
        //audioBuffer[2*k] = Re[k], 0<=k<n/2
        //audioBuffer[2*k+1] = Im[k], 0<k<n/2

        //Define the frequencies of interest
        float freqMin = 14400;
        float freqMax = 14500;

        //Loop through the fft bins and filter frequencies
        for(int fftBin = 0; fftBin < FFT_SIZE; fftBin++){        
            //Calculate the frequency of this bin assuming a sampling rate of 44,100 Hz
            float frequency = (float)fftBin * sampleRate / (float)FFT_SIZE;

            //Now filter the audio, I'm assuming you wanted to keep the
            //frequencies of interest rather than discard them.
            if(frequency  < freqMin || frequency > freqMax){
                //Calculate the index where the real and imaginary parts are stored
                int real = 2 * fftBin;
                int imaginary = 2 * fftBin + 1;

                //zero out this frequency
                audioBuffer[real] = 0;
                audioBuffer[imaginary] = 0;
            }
        }

        //Take the inverse FFT to convert signal from frequency to time domain
        mFFT.realInverse(audioBuffer, false);
    }

final AudioRecord findAudioRecord() {
        for (int rate : mSampleRates) {
            for (short audioFormat : new short[] { AudioFormat.ENCODING_PCM_8BIT, AudioFormat.ENCODING_PCM_16BIT }) {
                for (short channelConfig : new short[] { AudioFormat.CHANNEL_CONFIGURATION_MONO , AudioFormat.CHANNEL_CONFIGURATION_STEREO }) {
                    try {

                        bufferSize = AudioRecord.getMinBufferSize(rate, channelConfig, audioFormat);

                        if (bufferSize != AudioRecord.ERROR_BAD_VALUE) {
                            // check if we can instantiate and have a success
                            AudioRecord recorder = new AudioRecord(AudioSource.DEFAULT, rate, channelConfig, audioFormat, bufferSize);

                            if (recorder.getState() == AudioRecord.STATE_INITIALIZED){
                                Log.d(TAG, "Attempting rate " + rate + "Hz, bits: " + audioFormat + ", channel: "
                                        + channelConfig);
                                return recorder;
                            }
                        }
                    } catch (Exception e) {
                        Log.e(TAG, rate + "Exception, keep trying.",e);
                    }
                }
            }
        }

        return null;
    }

Answer 1

Q.1 Why we are converting “toTransform[i] = (double) buffer[i] / 32768.0; // signed 16 bit” buffer into 16bit value ?

32768 is the max value, I think that line of code is normalizing the data so that it is between 0 and 1.

Q.2 Right now audioRecord read data array is short array if i read byte array how will i convert it into singed 16 bit ?

Why are you reading it into a byte array? If you do that, then you have to combine two bytes together to get the 16 bit value. Instead just read into the short array to get the data. Here is an example of how to process the short array. This code reads it.

Q.3 I want to display sound frequency in Hz with double array how can i calculate sound frequency ?

This is one not-so-accurate way to estimate frequency. Other ways exists, but are more complex