How to read the data in a wav file to an array

Question

I need to get all the samples of a wav file into an array (or two if you need to do that to keep the stereo) so that I can apply some modifications to them. I was wondering if this is easily done (preferably without external libraries). I have no experience with reading in sound files, so I don't know much about the subject.

Answer 1

This code should do the trick. It converts a wave file to a normalized double array (-1 to 1), but it should be trivial to make it an int/short array instead (remove the /32768.0 bit and add 32768 instead). The right[] array will be set to null if the loaded wav file is found to be mono.

I can't claim it's completely bullet proof (potential off-by-one errors), but after creating a 65536 sample array, and creating a wave from -1 to 1, none of the samples appear to go 'through' the ceiling or floor.

// convert two bytes to one double in the range -1 to 1 static double bytesToDouble(byte firstByte, byte secondByte) {     // convert two bytes to one short (little endian)     short s = (secondByte << 8) | firstByte;     // convert to range from -1 to (just below) 1     return s / 32768.0; }  // Returns left and right double arrays. 'right' will be null if sound is mono. public void openWav(string filename, out double[] left, out double[] right) {     byte[] wav = File.ReadAllBytes(filename);      // Determine if mono or stereo     int channels = wav[22];     // Forget byte 23 as 99.999% of WAVs are 1 or 2 channels      // Get past all the other sub chunks to get to the data subchunk:     int pos = 12;   // First Subchunk ID from 12 to 16      // Keep iterating until we find the data chunk (i.e. 64 61 74 61 ...... (i.e. 100 97 116 97 in decimal))     while(!(wav[pos]==100 && wav[pos+1]==97 && wav[pos+2]==116 && wav[pos+3]==97)) {         pos += 4;         int chunkSize = wav[pos] + wav[pos + 1] * 256 + wav[pos + 2] * 65536 + wav[pos + 3] * 16777216;         pos += 4 + chunkSize;     }     pos += 8;      // Pos is now positioned to start of actual sound data.     int samples = (wav.Length - pos)/2;     // 2 bytes per sample (16 bit sound mono)     if (channels == 2) samples /= 2;        // 4 bytes per sample (16 bit stereo)      // Allocate memory (right will be null if only mono sound)     left = new double[samples];     if (channels == 2) right = new double[samples];     else right = null;      // Write to double array/s:     int i=0;     while (pos < length) {         left[i] = bytesToDouble(wav[pos], wav[pos + 1]);         pos += 2;         if (channels == 2) {             right[i] = bytesToDouble(wav[pos], wav[pos + 1]);             pos += 2;         }         i++;     } } 

Answer 2

Assuming your WAV file contains 16 bit PCM (which is the most common), you can use NAudio to read it out into a byte array, and then copy that into an array of 16 bit integers for convenience. If it is stereo, the samples will be interleaved left, right.

using (WaveFileReader reader = new WaveFileReader("myfile.wav")) {     Assert.AreEqual(16, reader.WaveFormat.BitsPerSample, "Only works with 16 bit audio");     byte[] buffer = new byte[reader.Length];     int read = reader.Read(buffer, 0, buffer.Length);     short[] sampleBuffer = new short[read / 2];     Buffer.BlockCopy(buffer, 0, sampleBuffer, 0, read); } 

I know you wanted to avoid third party libraries, but if you want to be sure to cope with WAV files with extra chunks, I suggest avoiding approaches like just seeking 44 bytes into the file.