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Downsampling and interpolation

I have audio files recorded at 44100 Hz, and I want to downsample them to 16000 Hz. I wrote a downsampler function that simply takes the factor of the original / desired and creates a new byte array of that size, and grabs the byte values using that factor as a jumping point.

downsample(byte[] inputAudio, inSample, outSample)

{

factor = inSample/ outSample;

length = inputAudio.length / factor;

newData = new byte[length];

           int index;
           for(int i=0; i<newData.length-1; i++)
           {
               index = (int)(i*factor);
               newData = audioData[index];
               i++;
               newData = audioData[index+1];
           }

//create wav header, output

}