Making Your Own Music Player: A Gentle Introduction to Audio Programming

July 15, 2017 ❖ Tags: tutorial, programming, audio, c

To start off, I'd like to say that I know very little about audio programming and digital audio in general. I've never formally studied signal processing, and hell, I haven't even started high school physics yet. This post merely documents what I've learned while trying to get sound working in my game, because there aren't really any other learning resources about this out there.

In this tutorial, we'll write a basic music player for Ogg Vorbis in C using two awesome libraries from Xiph.Org. The first, libao, will provide us with a means to play sound through our speakers, or headphones. or whatever, and we'll use libvorbisfile to decode the Ogg Vorbis files.

libao, like most other audio libraries, works by giving us a PCM buffer that we write sound data to, and that gets played back. PCM stands for Pulse-Code Modulation, and it's the basis of digital audio programming. You might have heard people talk about how analog audio is so much better than digital, and I think that learning the difference between the two helps to better understand digial audio. Historically, sound was recorded in terms of analog signals, which were easy to store as something like field strength on a magnetic medium. However, digitizing audio requires the signal to be either sampled or quantized. Both techniques are fairly similar, basically getting an instantaneous representation of the signal some number of times a second. The image below does a good job of explaining it, I think.

Figure 1: Diagram showing the difference between an analog sine wave and a quantization of that wave.

The rate at which the signal is sampled or quantized is the frequency. 44.1 kHz is typically the standard - meanining that 441,000 samples are taken every second. The number of channels is essentially how many speakers the sound is meant for. Stereo sound is the standard, so that is typically 2. And finally, the audio can be 8, 16, 24, or 32 bit, representing the size of the integer used to represent the sample.

Before we get into the code; you might need to configure libao if you're using PulseAudio. Just open it up in your favorite editor and change it as shown below.

$ sudo $EDITOR /etc/libao.conf
# Change from
# To
# Make sure to remove the dev=default line

Now we're ready to get into the code. We'll include the headers for libao and libvorbisfile, as well as some standard library headers and the size of the PCM buffer, which I'll explain soon.

#include <stdio.h>
#include <stdlib.h>

#include <ao/ao.h>
#include <vorbis/vorbisfile.h>

#define BUF_SIZE 256

The program is actually simple enough that we can do everything in main. For clarity, I'll be using C99 variable declaration. Our program will take the file to play as a command-line argument, so the first thing we need to do is check argc.

if (argc != 2) {
    fprintf(stderr, "Usage: %s [PATH]\n", argv[0]);
    return 1;

Next, we'll initialize libao. We'll also get the ID of the default sound driver for when we open an audio device later.

int default_driver = ao_default_driver_id();

Now, we'll specify the output format we want. This is what we were talking about earlier, about frequency and channels and such. The only part of this that wasn't mentioned was format.byte_format, which is just the byte order of the PCM buffer. The Vorbis decoder will work with either big or little endian, but we'll just stick with little endian for simplicity.

ao_sample_format format = {0};
format.bits        = 16;
format.channels    = 2;
format.rate        = 44100;
format.byte_format = AO_FMT_LITTLE;

We'll use this format structure to open an audio device with the default sound driver we figured out earlier.

ao_device *device = ao_open_live(default_driver, &format, NULL);
if (device == NULL) {
    fprintf(stderr, "Error opening device\n");
    return 1;

And now, we'll get our PCM buffer. Some audio libraries have a routine to give you a a buffer, but libao is alright with us using pretty much anything, so we'll allocate it with malloc(3). At this point, maybe you're wondering why we use a buffer. While we could read and play one byte at a time, that can be very inefficient. It's better to read it into a buffer, and then play that buffer. You don't want it to be too large, though, as there will be a longer pause every time the buffer has to be read into. You also don't want it to be too small. I find that 256 is good enough, but you can tweak that to your needs. The size should be a power of two.

char *buf = malloc(BUF_SIZE);
if (buf == NULL) {
    fprintf(stderr, "Error allocating PCM buffer.\n");
    return 1;

Now, we'll initialize libvorbisfile, which is done by opening the file we want to play. This huge switch statement isn't necessary, it's just there to show all the possible status codes of ov_fopen. Checking for a status code of 0 would be just fine here.

OggVorbis_File vf;
switch (ov_fopen(argv[1], &vf)) {
case OV_EREAD:
    fprintf(stderr, "Couldn't open %s.\n", argv[1]);
    return 1;

    fprintf(stderr, "File contains no vorbis data.\n");
    return 1;

    fprintf(stderr, "Vorbis version mismatch.\n");
    return 1;

    fprintf(stderr, "File contains a bad bitstream header.\n");
    return 1;

    fprintf(stderr, "Failure induced by heap/stack corruption.\n");
    return 1;

The real meat and potatoes of the program comes next. A loop that continually reads data into our PCM buffer and plays it, until there's no more data to play.

int read, bitstream;
do {
    read = ov_read(&vf, buf, BUF_SIZE, 0, 2, 1, &bitstream);
    ao_play(device, buf, BUF_SIZE);
} while (read > 0);

The random integer constants in the call to ov_read might be a bit intimidating, but it's really nothing to worry about. The first parameter is whether or not the PCM buffer is big endian (which it is not, so we pass 0), the second is the sample size, where 2 represents 16-bit, and the third is whether or not the data is signed. You can read more about it in the documentation.

Hopefully, things are starting to click around now. Any sound that comes out of your speakers is just a bunch of numbers, and file formats like Ogg and MP3 are just a means of compressing those numbers.

And finally, we'll finish up with some cleanup.

return 0;

Compilation is pretty easy, too.

$ gcc -o oggplay oggplay.c -lvorbisfile -lao

Pretty painless, right? Without error handling, this is about 21 lines of code.

Go ahead, try it out! If you don't save your music as Ogg Vorbis, you can convert songs with ffmpeg:

$ ffmpeg -i [file] -c:a libvorbis song.ogg

Here are some exercises if you want to play with this more:

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