Update, forgot snd_pcm_prepare(), now the problem is that the sound is
skipping.
#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
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;
snd_pcm_uframes_t frames_per_period = frames / PERIODS;
int dir = 0; // No idea what this does.
snd_pcm_hw_params_t *params;
int16_t framebuffer[FRAMEBUFFERSIZE * STEREO * BITS] = {0};
FILE *wav;
int size;
int status;
snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); //
SND_PCM_NONBLOCK);
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_buffer_size(handle, params, frames);
snd_pcm_hw_params_set_period_size(handle, params,
frames_per_period, dir);
snd_pcm_hw_params_set_rate(handle, params, SAMPLERATE, dir);
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_channels(handle, params, STEREO);
snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params(handle, params);
// Insert samples to framebuffer.
wav = fopen("pcm1611s.wav", "r");
fseek(wav, 0L, SEEK_END);
size = ftell(wav);
fseek(wav, 0L, SEEK_SET);
size /= sizeof(signed short);
while (size > 0)
{
fread(framebuffer, sizeof(int16_t), FRAMEBUFFERSIZE * STEREO *
BITS, wav);
status = snd_pcm_writei(handle, framebuffer, frames);
if (status == -EPIPE)
{
snd_pcm_prepare(handle);
}
size -= FRAMEBUFFERSIZE * STEREO * BITS;
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
exit(EXIT_SUCCESS);
}
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