ad_rec_t *ad_open_sps (int32 sps)
{
ad_rec_t *handle;
struct snd_pcm_channel_info info;
snd_pcm_t *dspH;
int err;
int card;
int dev;
if (sps != DEFAULT_SAMPLES_PER_SEC) {
if(abs(sps - DEFAULT_SAMPLES_PER_SEC) <= SPS_EPSILON) {
fprintf(stderr, "Audio sampling rate %d is within %d of %d samples/sec\n",
sps, SPS_EPSILON, DEFAULT_SAMPLES_PER_SEC);
} else {
fprintf(stderr, "Audio sampling rate %d not supported; must be %d samples/sec\n",
sps, DEFAULT_SAMPLES_PER_SEC);
return NULL;
}
}
card = 0;
dev = 0;
err = snd_pcm_open(&dspH, card, dev, SND_PCM_OPEN_DUPLEX); /* XXX */
if (err < 0) {
fprintf(stderr, "Error opening audio device %d,%din full-duplex: %s\n",
card, dev, snd_strerror(err));
return NULL;
}
err = snd_pcm_plugin_info(dspH, &info);
if (err < 0) {
fprintf(stderr, "Error getting PCM plugin info: %s\n", snd_strerror(err));
return NULL;
}
if (setparams(sps, dspH) < 0) {
return NULL;
}
if (setlevels(0, 0) < 0) {
return NULL;
}
if ((handle = (ad_rec_t *) calloc (1, sizeof(ad_rec_t))) == NULL) {
fprintf(stderr, "calloc(%d) failed\n", sizeof(ad_rec_t));
abort();
}
handle->dspH = dspH;
handle->recording = 0;
handle->sps = sps;
handle->bps = sizeof(int16);
return(handle);
}
ad_rec_t *
ad_open_dev(const char *dev, int32 sps)
{
ad_rec_t *handle;
snd_pcm_t *dspH;
int err;
if (dev == NULL)
dev = DEFAULT_DEVICE;
err = snd_pcm_open(&dspH, dev, SND_PCM_STREAM_CAPTURE, 0);
if (err < 0) {
fprintf(stderr,
"Error opening audio device %s for capture: %s\n",
dev, snd_strerror(err));
return NULL;
}
if (setparams(sps, dspH) < 0) {
return NULL;
}
if (setlevels(dev) < 0) {
return NULL;
}
if ((handle = (ad_rec_t *) calloc(1, sizeof(ad_rec_t))) == NULL) {
fprintf(stderr, "calloc(%d) failed\n", (int)sizeof(ad_rec_t));
abort();
}
handle->dspH = dspH;
handle->recording = 0;
handle->sps = sps;
handle->bps = sizeof(int16);
return (handle);
}
