static int
setup_recording(void)
{
int err;
if ((err = snd_pcm_capture_info(pcm_handle, &rec_info)) < 0) {
module_logger(&alsaSound, _L|LOG_USER | LOG_ERROR, _("record info error: %s\n"),
snd_strerror(err));
return 0;
}
/* Verify that our format is legal. */
if (rec_info.min_rate > rec_format.rate ||
rec_info.max_rate < rec_format.rate) {
module_logger(&alsaSound, _L|LOG_USER, _("device supports recording range %d to %d Hz"
"clipping default rate.\n"), rec_info.min_rate, rec_info.max_rate);
if (rec_info.min_rate > rec_format.rate)
rec_format.rate = rec_info.min_rate;
if (rec_format.rate > rec_info.max_rate)
rec_format.rate = rec_info.max_rate;
}
memset((void *) &rec_format, 0, sizeof(rec_format));
rec_format.rate = pcm_hz;
rec_format.channels = 1;
if (!(rec_info.formats & (1 << (rec_format.format = SND_PCM_SFMT_S8))) &&
!(rec_info.formats & (1 << (rec_format.format = SND_PCM_SFMT_U8)))) {
module_logger(&alsaSound, _L|LOG_ERROR | LOG_USER,
_("device does not support desired eight-bit recording format\n"));
//!!! until mix_server.c is more flexible
return 0;
}
if ((err = snd_pcm_capture_format(pcm_handle, &rec_format)) < 0) {
module_logger(&alsaSound, _L|LOG_ERROR | LOG_USER, _("device cannot set recording format\n"),
snd_strerror(err));
return 0;
}
/* Optimize fragments */
pcm_samplesize = 512;
samplebuf = (s16 *) malloc(pcm_samplesize * sizeof(samplebuf[0]));
// log(_("ALSA sample size = %d"), pcm_samplesize);
memset((void *) &rec_params, 0, sizeof(rec_params));
rec_params.fragment_size = pcm_samplesize * sizeof(s16);
if ((err = snd_pcm_capture_params(pcm_handle, &rec_params)) < 0) {
module_logger(&alsaSound, _L|LOG_ERROR | LOG_USER,
_("device cannot set requested PCM recording parameters:\n"
"%s\n"), snd_strerror(err));
return 0;
}
return 1;
}
static gboolean
alsa_open (void *dp)
{
alsa_driver * const d = dp;
int mf;
snd_pcm_format_t pf;
snd_pcm_capture_info_t pcm_info;
snd_pcm_capture_params_t pp;
snd_pcm_capture_status_t pbstat;
// int err = snd_pcm_open(&(d->soundfd), d->card_number, d->device_number,
// SND_PCM_OPEN_CAPTURE);
int err = snd_pcm_open(&(d->soundfd), 0, 0, SND_PCM_OPEN_CAPTURE);
if (err != 0) {
char buf[256];
g_sprintf(buf, _("Couldn't open ALSA device for sound input (card:%d, device:%d):\n%s"),
d->card_number, d->device_number, snd_strerror(err));
error_error(buf);
goto out;
}
// ---
// Set non-blocking mode.
// ---
snd_pcm_block_mode(d->soundfd, 1); // enable block mode
d->bits = 0;
mf = 0;
// ---
// Select audio format
// ---
memset(&pf, 0, sizeof(pf));
if (d->p_resolution == 16) {
pf.format = SND_PCM_SFMT_S16_LE;
d->bits = 16;
mf = ST_MIXER_FORMAT_S16_LE;
}
else {
pf.format = SND_PCM_SFMT_U8;
d->bits = 8;
mf = ST_MIXER_FORMAT_S8;
}
/* if(d->p_channels == 2) { */
/* d->stereo = 1; */
/* pf.channels = d->p_channels; */
/* mf |= ST_MIXER_FORMAT_STEREO; */
/* } */
/* else { */
/* pf.channels = d->p_channels; */
/* d->stereo = 0; */
/* } */
d->stereo = 0;
pf.channels = 1;
pf.rate = d->p_mixfreq;
d->playrate = d->p_mixfreq;
mf = ST_MIXER_FORMAT_S16_LE;
d->mf = mf;
err = snd_pcm_capture_format(d->soundfd, &pf);
if (err < 0) {
error_error(_("Required sound output format not supported.\n"));
goto out;
}
snd_pcm_capture_info(d->soundfd, &pcm_info);
memset(&pp, 0, sizeof(pp));
pp.fragment_size = d->p_fragsize * pf.channels * d->bits / 8;
pp.fragments_min = 1;
err = snd_pcm_capture_params(d->soundfd, &pp);
if (err < 0) {
error_error(_("Required sound output parameters not supported.\n"));
goto out;
}
snd_pcm_capture_status(d->soundfd, &pbstat);
/* d->numfrags = pbstat.fragments; */
d->numfrags = 1;
d->fragsize = pbstat.fragment_size;
d->sndbuf = calloc(1, d->fragsize);
if(d->stereo == 1) {
d->fragsize /= 2;
}
if(d->bits == 16) {
d->fragsize /= 2;
}
d->polltag = gdk_input_add(snd_pcm_file_descriptor(d->soundfd), GDK_INPUT_READ, alsa_poll_ready_sampling, d);
// d->firstpoll = TRUE;
// d->playtime = 0;
return TRUE;
out:
alsa_release(dp);
return FALSE;
}
