How do I generate an audio sine or square wave of a given frequency?
I am hoping to do this to calibrate equipment, so how precise would these waves be?
A square wave can be created by adding the sum of the odd harmonics of a sine wave. This is better explained for odd harmonics as follows. For the LLC, a higher switching frequency of 90 kHz is used in place of the 50Hz fundamental frequency in the aforementioned example.
The general formula for a good sine function is y=Asin(2πfx+B), where A is the amplitude, f is the frequency, and B is the phase.
Sine wave inverters are more efficient than square wave inverters, when it comes to the conversion of DC to AC. This ensures that the power loss is minimised, due to greater efficiency. As a consequence, your electricity bill does not shoot up. This is an immediate benefit of using a sine wave inverter.
A popular way to make a sine wave oscillator is to use an RC network to produce a 180 degree phase shift to use in the feedback path of an inverting amplifier. Setting the gain of the amplifier to offset the RC network attenuation will produce oscillation.
This lets you give frequency, duration, and amplitude, and it is 100% .NET CLR code. No external DLL's. It works by creating a WAV-formatted MemoryStream
which is like creating a file in memory only, without storing it to disk. Then it plays that MemoryStream
with System.Media.SoundPlayer
.
using System;
using System.Collections.Generic;
using System.IO;
using System.Linq;
using System.Windows.Forms;
public static void PlayBeep(UInt16 frequency, int msDuration, UInt16 volume = 16383)
{
var mStrm = new MemoryStream();
BinaryWriter writer = new BinaryWriter(mStrm);
const double TAU = 2 * Math.PI;
int formatChunkSize = 16;
int headerSize = 8;
short formatType = 1;
short tracks = 1;
int samplesPerSecond = 44100;
short bitsPerSample = 16;
short frameSize = (short)(tracks * ((bitsPerSample + 7) / 8));
int bytesPerSecond = samplesPerSecond * frameSize;
int waveSize = 4;
int samples = (int)((decimal)samplesPerSecond * msDuration / 1000);
int dataChunkSize = samples * frameSize;
int fileSize = waveSize + headerSize + formatChunkSize + headerSize + dataChunkSize;
// var encoding = new System.Text.UTF8Encoding();
writer.Write(0x46464952); // = encoding.GetBytes("RIFF")
writer.Write(fileSize);
writer.Write(0x45564157); // = encoding.GetBytes("WAVE")
writer.Write(0x20746D66); // = encoding.GetBytes("fmt ")
writer.Write(formatChunkSize);
writer.Write(formatType);
writer.Write(tracks);
writer.Write(samplesPerSecond);
writer.Write(bytesPerSecond);
writer.Write(frameSize);
writer.Write(bitsPerSample);
writer.Write(0x61746164); // = encoding.GetBytes("data")
writer.Write(dataChunkSize);
{
double theta = frequency * TAU / (double)samplesPerSecond;
// 'volume' is UInt16 with range 0 thru Uint16.MaxValue ( = 65 535)
// we need 'amp' to have the range of 0 thru Int16.MaxValue ( = 32 767)
double amp = volume >> 2; // so we simply set amp = volume / 2
for (int step = 0; step < samples; step++)
{
short s = (short)(amp * Math.Sin(theta * (double)step));
writer.Write(s);
}
}
mStrm.Seek(0, SeekOrigin.Begin);
new System.Media.SoundPlayer(mStrm).Play();
writer.Close();
mStrm.Close();
} // public static void PlayBeep(UInt16 frequency, int msDuration, UInt16 volume = 16383)
You can use NAudio and create a derived WaveStream that outputs sine or square waves which you could output to the soundcard or write to a WAV file. If you used 32-bit floating point samples you could write the values directly out of the sin function without having to scale as it already goes between -1 and 1.
As for accuracy, do you mean exactly the right frequency, or exactly the right wave shape? There is no such thing as a true square wave, and even the sine wave will likely have a few very quiet artifacts at other frequencies. If it's accuracy of frequency that matters, you are reliant on the stability and accuracy of the clock in your sound card. Having said that, I would imagine that the accuracy would be good enough for most uses.
Here's some example code that makes a 1 kHz sample at a 8 kHz sample rate and with 16 bit samples (that is, not floating point):
int sampleRate = 8000;
short[] buffer = new short[8000];
double amplitude = 0.25 * short.MaxValue;
double frequency = 1000;
for (int n = 0; n < buffer.Length; n++)
{
buffer[n] = (short)(amplitude * Math.Sin((2 * Math.PI * n * frequency) / sampleRate));
}
Try from Creating sine and save to wave file in C#
private void TestSine()
{
IntPtr format;
byte[] data;
GetSineWave(1000, 100, 44100, -1, out format, out data);
WaveWriter ww = new WaveWriter(File.Create(@"d:\work\sine.wav"),
AudioCompressionManager.FormatBytes(format));
ww.WriteData(data);
ww.Close();
}
private void GetSineWave(double freq, int durationMs, int sampleRate, short decibel, out IntPtr format, out byte[] data)
{
short max = dB2Short(decibel);//short.MaxValue
double fs = sampleRate; // sample freq
int len = sampleRate * durationMs / 1000;
short[] data16Bit = new short[len];
for (int i = 0; i < len; i++)
{
double t = (double)i / fs; // current time
data16Bit[i] = (short)(Math.Sin(2 * Math.PI * t * freq) * max);
}
IntPtr format1 = AudioCompressionManager.GetPcmFormat(1, 16, (int)fs);
byte[] data1 = new byte[data16Bit.Length * 2];
Buffer.BlockCopy(data16Bit, 0, data1, 0, data1.Length);
format = format1;
data = data1;
}
private static short dB2Short(double dB)
{
double times = Math.Pow(10, dB / 10);
return (short)(short.MaxValue * times);
}
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