Right now i have an audio file (2 Channels, 44.1kHz Sample Rate, 16bit Sample size, WAV) I would like to pass it into this method but i am not sure of any way to convert the WAV file to a byte array.
/// <summary>
/// Process 16 bit sample
/// </summary>
/// <param name="wave"></param>
public void Process(ref byte[] wave)
{
_waveLeft = new double[wave.Length / 4];
_waveRight = new double[wave.Length / 4];
if (_isTest == false)
{
// Split out channels from sample
int h = 0;
for (int i = 0; i < wave.Length; i += 4)
{
_waveLeft[h] = (double)BitConverter.ToInt16(wave, i);
_waveRight[h] = (double)BitConverter.ToInt16(wave, i + 2);
h++;
}
}
else
{
// Generate artificial sample for testing
_signalGenerator = new SignalGenerator();
_signalGenerator.SetWaveform("Sine");
_signalGenerator.SetSamplingRate(44100);
_signalGenerator.SetSamples(16384);
_signalGenerator.SetFrequency(5000);
_signalGenerator.SetAmplitude(32768);
_waveLeft = _signalGenerator.GenerateSignal();
_waveRight = _signalGenerator.GenerateSignal();
}
// Generate frequency domain data in decibels
_fftLeft = FourierTransform.FFTDb(ref _waveLeft);
_fftRight = FourierTransform.FFTDb(ref _waveRight);
}
Edit Hi sorry for the confusion. I'm currently new to audio signalling so my explanation of what I might like to get is wrong. For this method to work correctly, i believe i need to pass in the byte array of the data chunk in the wav file only. The end result would be to apply fft on it as shown in the code and transform it to a spectrogram. Thanks.