Audio signal analysis to get information about frequencies.

SignalAnalysis.cs

using System;
using NAudio.Dsp;
using NAudio.Wave;
using NAudio.Wave.SampleProviders;

namespace GM.SP.Audio
{
	public static class SignalAnalysis
	{
		public class Frequency
		{
			public float Hz { get; set; }
			public float Amplitude { get; set; }
			public float Phase { get; set; }
		}

		public static Frequency[] AnalyzeAudioFile(string audioFile)
		{
			WaveStream audioWaveProvider = null;

			try {
				// open audio file
				if(audioFile.ToLower().EndsWith(".mp3"))
					audioWaveProvider = new Mp3FileReader(audioFile);
				else if(audioFile.ToLower().EndsWith(".wav"))
					audioWaveProvider = new WaveFileReader(audioFile);
				else
					throw new Exception("Unsupported format.");

				// create a sample provider
				ISampleProvider audioSampleProvider;
				switch(audioWaveProvider.WaveFormat.BitsPerSample) {
					case 8:
						audioSampleProvider = new Pcm8BitToSampleProvider(audioWaveProvider);
						break;
					case 16:
						audioSampleProvider = new Pcm16BitToSampleProvider(audioWaveProvider);
						break;
					case 24:
						audioSampleProvider = new Pcm24BitToSampleProvider(audioWaveProvider);
						break;
					case 32:
						audioSampleProvider = new Pcm32BitToSampleProvider(audioWaveProvider);
						break;
					default:
						throw new Exception("Unsupported bits per sample in audio file.");
				}

				// read all samples (includes left and right channel in case of stereo)
				float[] allSamples = new float[audioWaveProvider.Length / (audioWaveProvider.WaveFormat.BitsPerSample / 8)];
				audioSampleProvider.Read(allSamples, 0, allSamples.Length);

				// unzip to one channel samples that are to be analyzed
				float[] samples;
				if(audioWaveProvider.WaveFormat.Channels == 2)
					samples = AudioUtility.Unzip(allSamples).Item1;
				else
					samples = allSamples;

				return AnalyzeAudio(samples,audioWaveProvider.WaveFormat.SampleRate);
			} finally {
				audioWaveProvider?.Dispose();
			}
		}

		/// <summary>
		/// If the count of samples is not a power of 2, it will be padded by zeros until the next larger power of 2.
		/// </summary>
		public static Frequency[] AnalyzeAudio(float[] samples,int sampleRate)
		{
			// size must be a power of 2
			int size = 1;
			int m = 0;
			while(size < samples.Length) {
				size <<= 1;
				++m;
			}

			// create complex numbers from floats (X is real, Y is imag)
			Complex[] fftResults = new Complex[size];
			for(int i = samples.Length - 1; i >= 0; --i)
				fftResults[i] = new Complex() { X = samples[i], Y = 0 };

			// fourier transform
			FastFourierTransform.FFT(true, m, fftResults);

			float frequencyStep = sampleRate / (float)size;

			Frequency[] frequencies = new Frequency[size / 2];
			for(int i = 0; i < frequencies.Length; ++i) {
				Complex v = fftResults[i];

				Frequency f = new Frequency();
				f.Hz = (i+1) * frequencyStep;
				f.Amplitude = (float)Math.Sqrt(Math.Pow(v.X, 2) + Math.Pow(v.Y, 2));
				f.Phase = (float)Math.Atan(v.Y / v.X);

				frequencies[i] = f;
			}
			
			return frequencies;
		}
	}
}