Why does the sound skip in ALSA?

Question

I am trying to play this sound using the program below, but the sound is a little fast and it skips. The sound file's sample rate is 11025 Hz, stereo, the sample size is 16 bit. The issue appears to be snd_pcm_hw_params_set_buffer_size() and snd_pcm_hw_params_set_period_size() (both are functions I am studying) due to invalid arguments, but I don't know why they are invalid and commenting them out doesn't address the skipping effect. What am I doing wrong?

#include <stdlib.h>
#include <stdint.h>
#include <alsa/asoundlib.h>

#define STEREO              2
#define BITS            16 / 8
#define FRAMEBUFFERSIZE   512 // in samples
#define PERIODS             2
#define SAMPLERATE      11025

void snd_error_checker(int error, char *function_name)
{
    if (error)
    {
        printf("Error (%s): %s\n", function_name, snd_strerror(error));
        // exit(EXIT_FAILURE);
    }
}

int main(void)
{
    snd_pcm_t *handle;
    uint32_t channels                   = STEREO;
    uint32_t sample_size                = BITS; // 16 bit
    uint32_t frame_size                 = sample_size * channels;
    snd_pcm_uframes_t frames            = FRAMEBUFFERSIZE / frame_size;
    snd_pcm_uframes_t frames_per_period = frames / PERIODS;
    int32_t dir = 0; // No idea what this does.
    snd_pcm_hw_params_t *params;
    int16_t *buffer;
    FILE *wav;
    int32_t size;
    int error;

    error = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
    snd_error_checker(error, "snd_pcm_open()");
    snd_pcm_hw_params_alloca(&params);
    error = snd_pcm_hw_params_any(handle, params);
    snd_error_checker(error, "snd_pcm_hw_params_any()");
    error = snd_pcm_hw_params_set_buffer_size(handle, params, frames);
    snd_error_checker(error, "snd_pcm_hw_params_set_buffer_size()");
    error = snd_pcm_hw_params_set_period_size(handle, params, frames_per_period, dir);
    snd_error_checker(error, "snd_pcm_hw_params_set_period_size()");
    error = snd_pcm_hw_params_set_rate(handle, params, SAMPLERATE, dir);
    snd_error_checker(error, "snd_pcm_hw_params_set_rate()");
    error = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
    snd_error_checker(error, "snd_pcm_hw_params_set_access()");
    error = snd_pcm_hw_params_set_channels(handle, params, STEREO);
    snd_error_checker(error, "snd_pcm_hw_params_set_channels()");
    error = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
    snd_error_checker(error, "snd_pcm_hw_params_set_format()");
    error = snd_pcm_hw_params(handle, params);
    snd_error_checker(error, "snd_pcm_hw_params()");

    // Insert samples to framebuffer.
    wav = fopen("pcm1611s.wav", "r");
    fseek(wav, 0L, SEEK_END);
    size = ftell(wav);
    fseek(wav, 0L, SEEK_SET);
    buffer = malloc(size);
    fread(buffer, sizeof(int16_t), size, wav);
    fclose(wav);
    size /= sizeof(int16_t);

    // Set ptrbuffer 46 bytes ahead to skip the header.
    for (int16_t *ptrbuffer = buffer + 46; size > ptrbuffer - buffer; ptrbuffer += FRAMEBUFFERSIZE * STEREO * BITS)
    {
        error = snd_pcm_writei(handle, ptrbuffer, frames);
        if (error)
        {
            snd_pcm_recover(handle, error, 0);
        }
    }
    snd_pcm_drain(handle);
    snd_pcm_close(handle);
    exit(EXIT_SUCCESS);
}

Answer 1

#define FRAMEBUFFERSIZE   512 // in samples

The comment is lying; the value is used as a byte count.

uint32_t sample_size                = BITS; // 16 bit

The comment is lying; the value is actualy measured in bytes. (And BITS is named wrong.)

fread(buffer, sizeof(int16_t), size, wav);

size is measured in bytes, but you tell fread() to read 16-bit words. (This does not actually hurt because it stops reading at the end of the file. But you should have checked for errors.)

int16_t *ptrbuffer = ...

You are telling the compiler that ptrbuffer points to 16-bit values.

ptrbuffer += FRAMEBUFFERSIZE * STEREO * BITS

You are telling the compiler to step over 2048 16-bit values, i.e., over 4096 bytes. You actually want to step over all the sample in a buffer, that is, frames * STEREO.

error = snd_pcm_writei(...);
if (error)

snd_pcm_writei() returns a positive number on success; errors are negative.