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(¶ms);
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);
}
#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.