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Home/ Questions/Q 6956237
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Editorial Team
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Editorial Team
Asked: May 27, 20262026-05-27T14:50:34+00:00 2026-05-27T14:50:34+00:00

I am using DirectSound to write a sine wave to the audio card. The

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I am using DirectSound to write a sine wave to the audio card. The sample-size is 16-bit, one channel. My question is, how many samples should it take to make a five-second sound? The sample rate is 44100 samples per second. The math is simple: 220500 is the answer. This is driving me crazy, though, because my code plays only roughly half that time!! Here’s my code:

using Microsoft.DirectX.DirectSound; 
using System;
namespace Audio
{
    // The class 
    public class Oscillator
    {
        static void Main(string[] args)
        {

            // Set up wave format 
            WaveFormat waveFormat = new WaveFormat();
            waveFormat.FormatTag = WaveFormatTag.Pcm;
            waveFormat.Channels = 1;
            waveFormat.BitsPerSample = 16;
            waveFormat.SamplesPerSecond = 44100;
            waveFormat.BlockAlign = (short)(waveFormat.Channels * waveFormat.BitsPerSample / 8);
            waveFormat.AverageBytesPerSecond = waveFormat.BlockAlign * waveFormat.SamplesPerSecond;

            // Set up buffer description 
            BufferDescription bufferDesc = new BufferDescription(waveFormat);
            bufferDesc.Control3D = false;
            bufferDesc.ControlEffects = false;
            bufferDesc.ControlFrequency = true;
            bufferDesc.ControlPan = true;
            bufferDesc.ControlVolume = true;
            bufferDesc.DeferLocation = true;
            bufferDesc.GlobalFocus = true;

            Device d = new Device();
            d.SetCooperativeLevel(new System.Windows.Forms.Control(), CooperativeLevel.Priority);


            int samples = 5 * waveFormat.SamplesPerSecond * waveFormat.Channels;
            char[] buffer = new char[samples];

            // Set buffer length 
            bufferDesc.BufferBytes = buffer.Length * waveFormat.BlockAlign;

            // Set initial amplitude and frequency 
            double frequency = 500;
            double amplitude = short.MaxValue / 3;
            double two_pi = 2 * Math.PI;
            // Iterate through time 
            for (int i = 0; i < buffer.Length; i++)
            {
                // Add to sine 
                buffer[i] = (char)(amplitude *
                    Math.Sin(i * two_pi * frequency / waveFormat.SamplesPerSecond));
            }

            SecondaryBuffer bufferSound = new SecondaryBuffer(bufferDesc, d);
            bufferSound.Volume = (int)Volume.Max;
            bufferSound.Write(0, buffer, LockFlag.None);
            bufferSound.Play(0, BufferPlayFlags.Default);
            System.Threading.Thread.Sleep(10000);
        }
    }
}

By my calculations, this should play for 5 seconds. It plays for half-time. If I change

 int samples = 5 * waveFormat.SamplesPerSecond * waveFormat.Channels;

to

  int samples = 5 * waveFormat.SamplesPerSecond * waveFormat.Channels
      * waveFormat.BlockAlign;

Then the sound works alright, but that’s an hack, right? Surely I’m doing something wrong, but I can’t tell what.

Thanks for your time.

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1 Answer

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  1. Editorial Team
    Editorial Team
    2026-05-27T14:50:35+00:00Added an answer on May 27, 2026 at 2:50 pm

    You will have 2 bytes per sample for 16-bit if I’m not mistaken, therefore your buffers byte count will be twice the sample count.

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