If you want to work at an even lower level of abstraction, a good environment for experimentation is the Linux operating system using “from scratch” programs written in C++. In our first example C++ sound program, we show you how to create sound waves of a given frequency, add frequency components to get a complex wave, and play the sounds via the sound device. This program is another implementation of the exercises in Max and MATLAB in Sections 2.3.1 and 2.3.3.   The C++ program is given in Program 2.4.  

Aside:  In this example program, 8 bits are used to store each audio sample.  That is, the bit depth is 8. The sound library also allows a bit depth of 16.  The concept of bit depth will be explained in detail in Chapter 5.


//This program uses the OSS library.
#include <sys/ioctl.h> //for ioctl()
#include <math.h> //sin(), floor(), and pow()
#include <stdio.h> //perror
#include <fcntl.h> //open, O_WRONLY
#include <linux/soundcard.h> //SOUND_PCM*
#include <iostream>
#include <unistd.h>
using namespace std;

#define TYPE char
#define LENGTH 1 //number of seconds per frequency
#define RATE 44100 //sampling rate
#define SIZE sizeof(TYPE) //size of sample, in bytes
#define CHANNELS 1 //number of audio channels
#define PI 3.14159
#define NUM_FREQS 3 //total number of frequencies
#define BUFFSIZE (int) (NUM_FREQS*LENGTH*RATE*SIZE*CHANNELS) //bytes sent to audio device
#define ARRAYSIZE (int) (NUM_FREQS*LENGTH*RATE*CHANNELS) //total number of samples
#define SAMPLE_MAX (pow(2,SIZE*8 - 1) - 1) 

void writeToSoundDevice(TYPE buf[], int deviceID) {
	int status;
	status = write(deviceID, buf, BUFFSIZE);
	if (status != BUFFSIZE)
		perror("Wrote wrong number of bytes\n");
	status = ioctl(deviceID, SNDCTL_DSP_SYNC, 0);
	if (status == -1)
		perror("SNDCTL_DSP_SYNC failed\n");

int main() {
	int deviceID, arg, status, f, t, a, i;
	deviceID = open("/dev/dsp", O_WRONLY, 0);
	if (deviceID < 0)
		perror("Opening /dev/dsp failed\n");
// working
	arg = SIZE * 8;
	status = ioctl(deviceID, SNDCTL_DSP_SETFMT, &arg);
	if (status == -1)
		perror("Unable to set sample size\n");
	arg = CHANNELS;
	status = ioctl(deviceID, SNDCTL_DSP_CHANNELS, &arg);
	if (status == -1)
		perror("Unable to set number of channels\n");
	arg = RATE;
	status = ioctl(deviceID, SNDCTL_DSP_SPEED, &arg);
	if (status == -1)
		perror("Unable to set sampling rate\n");
	for (i = 0; i < NUM_FREQS; ++i) {
		switch (i) {
			case 0:
				f = 262;
			case 1:
				f = 330;
			case 2:
				f = 392;
		for (t = 0; t < ARRAYSIZE/NUM_FREQS; ++t) {
			buf[t + ((ARRAYSIZE / NUM_FREQS) * i)] = floor(a * sin(2*PI*f*t/RATE));
	writeToSoundDevice(buf, deviceID);

Program 2.4 Adding sine waves and sending sound to sound device in C++

To be able to compile and run a program such as this, you need to install a sound library in your Linux environment. At the time of the writing of this chapter, the two standard low-level sound libraries for Linux are the OSS (Open Sound System) and ALSA (Advanced Linux Sound Architecture). A sound library provides a software interface that allows your program to access the sound devices, sending and receiving sound data. ALSA is the newer of the two libraries and is preferred by most users. At a slightly higher level of abstraction are PulseAudio and Jack, applications which direct multiple sound streams from their inputs to their outputs. Ultimately, PulseAudio and Jack use lower level libraries to communicate directly with the sound cards.

Program 2.4 uses the OSS library. In a program such as this, the sound device is opened, read from, and written to in a way similar to how files are handled. The sample program shows how you open /dev/dsp, an interface to the sound card device, to ask this device to receive audio data. The variable deviceID serves as an ID of the sound device and is used as a parameter indicating the size of data to expect, the number of channels, and the data rate. We’ve set a size of eight bits (one byte) per audio sample, one channel, and a data rate of 44,100 samples per second. The significance of these numbers will be clearer when we talk about digitization in Chapter 5. The buffer size is a product of the sample size, data rate, and length of the recording (in this case, three seconds), yielding a buffer of 44,100 * 3 bytes.

The sound wave is created by taking the sine of the appropriate frequency (262 Hz, for example) at 44,100 evenly-spaced intervals for one second of audio data. The value returned from the sine function is between -1 and 1. However, the sound card expects a value that is stored in one byte (i.e., 8 bits), ranging from -128 to 127. To put the value into this range, we multiply by 127 and, with the floor function, round down.

The three frequencies are created and concatenated into one array of audio values. The write function has the device ID, the name of the buffer for storing the sound data, and the size of the buffer as its parameters. This function sends the sound data to the sound card to be played. The three frequencies together produce a harmonious chord in the key of C. In Chapter 3, we’ll explore what makes these frequencies harmonious.

The program requires some header files for definitions of constants like O_WRONLY (restricting access to the sound device to writing) and SOUND_PCM_WRITE_BITS. After you install the sound libraries, you’ll need to locate the appropriate header files and adjust the #include statement accordingly. You’ll also need to check the way your compiler handles the math and sound libraries. You may need to include the option –lm on the compile line to include the math library, or the –lasound option for the ALSA library.

This program introduces you to the notion that sound must be converted to a numeric format that is communicable to a computer. The solution to the programming assignment given as a learning supplement has an explanation of the variables and constants in this program. A full understanding of the program requires that you know something about sampling and quantization, the two main steps in analog-to-digital conversion, a topic that we’ll examine in depth in Chapter 5.