static int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams)
{
int err;
/* get the current swparams */
err = alsa_pcm_sw_params_current(handle, swparams);
if (err < 0) {
printf("Unable to determine current swparams for playback: %s\n", alsa_strerror(err));
return err;
}
/* start the transfer when the buffer is almost full: */
/* (buffer_size / avail_min) * avail_min */
err = alsa_pcm_sw_params_set_start_threshold(handle, swparams, (buffer_size_in_frames / period_size) * period_size);
if (err < 0) {
printf("Unable to set start threshold mode for playback: %s\n", alsa_strerror(err));
return err;
}
/* allow the transfer when at least period_size samples can be processed */
/* or disable this mechanism when period event is enabled (aka interrupt like style processing) */
err = alsa_pcm_sw_params_set_avail_min(handle, swparams, period_size);
if (err < 0) {
printf("Unable to set avail min for playback: %s\n", alsa_strerror(err));
return err;
}
/* write the parameters to the playback device */
err = alsa_pcm_sw_params(handle, swparams);
if (err < 0) {
printf("Unable to set sw params for playback: %s\n", alsa_strerror(err));
return err;
}
return 0;
}
static void ALSA_Update(void)
{
int err;
if (!enabled) return;
if (bytes_written == 0 || bytes_played == bytes_written) {
bytes_written = VC_WriteBytes(audiobuffer,period_size * global_frame_size);
bytes_played = 0;
}
while (bytes_played < bytes_written)
{
err = alsa_pcm_writei(pcm_h, &audiobuffer[bytes_played], (bytes_written - bytes_played) / global_frame_size);
if (err == -EAGAIN)
continue;
if (err < 0) {
if ((err = xrun_recovery(pcm_h, err)) < 0) {
_mm_errno = MMERR_ALSA_PCM_RECOVER;
enabled = 0;
dbgprint(stderr, "Write error: %s\n", alsa_strerror(err));
}
break;
}
bytes_played += err * global_frame_size;
}
}
static void ALSA_Update(void)
{
int err;
{
if (bytes_written == 0 || bytes_played == bytes_written)
{
bytes_written = VC_WriteBytes(audiobuffer,buffer_size_in_frames * global_frame_size);
bytes_played = 0;
}
while (bytes_played < bytes_written)
{
err = alsa_pcm_writei(pcm_h, &audiobuffer[bytes_played], (bytes_written - bytes_played) / global_frame_size);
if (err == -EAGAIN)
{
continue;
}
if (err < 0) {
if ((err = xrun_recovery(pcm_h, err)) < 0) {
printf("Write error: %s\n", alsa_strerror(err));
exit(-1);
}
break;
}
bytes_played += err * global_frame_size;
}
}
}
static BOOL ALSA_Init_internal(void)
{
snd_pcm_format_t pformat;
#ifdef OLD_ALSA
#define channels pformat.channels
#define rate pformat.rate
int mask,card;
#else
int rate;
int channels;
int err;
snd_pcm_hw_params_t * hwparams;
snd_pcm_sw_params_t * swparams;
#endif
/* adjust user-configurable settings */
if((getenv("MM_NUMFRAGS"))&&(numfrags==DEFAULT_NUMFRAGS)) {
numfrags=atoi(getenv("MM_NUMFRAGS"));
if ((numfrags<2)||(numfrags>16)) numfrags=DEFAULT_NUMFRAGS;
}
#ifdef OLD_ALSA
if((getenv("ALSA_CARD"))&&(!cardmin)&&(cardmax==SND_CARDS)) {
cardmin=atoi(getenv("ALSA_CARD"));
cardmax=cardmin+1;
#endif
if(getenv("ALSA_PCM"))
device=atoi(getenv("ALSA_PCM"));
#ifdef OLD_ALSA
}
#endif
/* setup playback format structure */
#define NUM_CHANNELS() ((md_mode&DMODE_STEREO)?2:1)
#ifdef OLD_ALSA
memset(&pformat,0,sizeof(pformat));
#ifdef SND_LITTLE_ENDIAN
pformat.format=(md_mode&DMODE_16BITS)?SND_PCM_SFMT_S16_LE:SND_PCM_SFMT_U8;
#else
pformat.format=(md_mode&DMODE_16BITS)?SND_PCM_SFMT_S16_BE:SND_PCM_SFMT_U8;
#endif
#else
pformat = (md_mode&DMODE_16BITS)?SND_PCM_FORMAT_S16_LE:SND_PCM_FORMAT_U8;
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
#endif
channels = NUM_CHANNELS();
rate=md_mixfreq;
/* scan for appropriate sound card */
#ifdef OLD_ALSA
mask=alsa_cards_mask();
#endif
_mm_errno=MMERR_OPENING_AUDIO;
#ifdef OLD_ALSA
for (card=cardmin;card<cardmax;card++)
#endif
{
#ifdef OLD_ALSA
struct snd_ctl_hw_info info;
snd_ctl_t *ctl_h;
int dev,devmin,devmax;
#endif
#ifdef OLD_ALSA
/* no card here, onto the next */
if (!(mask&(1<<card))) continue;
#endif
#ifdef OLD_ALSA
/* try to open the card in query mode */
if (alsa_ctl_open(&ctl_h,card)<0)
continue;
#else
if ((err = alsa_pcm_open(&pcm_h, "plughw:0,0", SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
printf("snd_pcm_open() call failed: %s\n", alsa_strerror(err));
goto END;
}
#endif
#ifdef OLD_ALSA
/* get hardware information */
if(alsa_ctl_hw_info(ctl_h,&info)<0) {
alsa_ctl_close(ctl_h);
continue;
}
/* scan subdevices */
if(device==-1) {
devmin=0;devmax=info.pcmdevs;
} else
devmin=devmax=device;
#endif
#ifdef OLD_ALSA
for(dev=devmin;dev<devmax;dev++)
#endif
{
#ifdef OLD_ALSA
int size,bps;
snd_pcm_info_t pcminfo;
snd_pcm_playback_info_t ctlinfo;
struct snd_pcm_playback_info pinfo;
struct snd_pcm_playback_params pparams;
/* get PCM capabilities */
if(alsa_ctl_pcm_info(ctl_h,dev,&pcminfo)<0)
continue;
/* look for playback capability */
if(!(pcminfo.flags&SND_PCM_INFO_PLAYBACK))
continue;
/* get playback information */
#if defined(SND_LIB_VERSION) && (SND_LIB_VERSION >= 0x400)
if(alsa_ctl_pcm_playback_info(ctl_h,dev,0,&ctlinfo)<0)
continue;
#else
if(alsa_ctl_pcm_playback_info(ctl_h,dev,&ctlinfo)<0)
continue;
#endif
/*
If control goes here, we have found a sound device able to play PCM data.
Let's open in in playback mode and see if we have compatible playback
settings.
*/
if (alsa_pcm_open(&pcm_h,card,dev,SND_PCM_OPEN_PLAYBACK)<0)
continue;
if (alsa_pcm_playback_info(pcm_h,&pinfo)<0) {
alsa_pcm_close(pcm_h);
pcm_h=NULL;
continue;
}
/* check we have compatible settings */
if((pinfo.min_rate>rate)||(pinfo.max_rate<rate)||
(!(pinfo.formats&(1<<pformat.format)))) {
alsa_pcm_close(pcm_h);
pcm_h=NULL;
continue;
}
fragmentsize=pinfo.buffer_size/numfrags;
#ifdef MIKMOD_DEBUG
if ((fragmentsize<512)||(fragmentsize>16777216L))
fprintf(stderr,"\rweird pinfo.buffer_size:%d\n",pinfo.buffer_size);
#endif
alsa_pcm_flush_playback(pcm_h);
/* set new parameters */
if(alsa_pcm_playback_format(pcm_h,&pformat)<0)
#else
if( alsa_pcm_set_params(pcm_h, pformat,
SND_PCM_ACCESS_RW_INTERLEAVED,
channels,
rate,
1,
500000 /* 0.5sec */
) < 0)
#endif
{
alsa_pcm_close(pcm_h);
pcm_h=NULL;
#ifdef OLD_ALSA
continue;
#else
goto END;
#endif
}
global_frame_size = channels*(md_mode&DMODE_16BITS?2:1);
#ifdef OLD_ALSA
/* compute a fragmentsize hint
each fragment should be shorter than, but close to, half a
second of playback */
bps=(rate*global_frame_size)>>1;
size=fragmentsize;while (size>bps) size>>=1;
#endif
#ifdef MIKMOD_DEBUG
if (size < 16) {
fprintf(stderr,"\rweird hint result:%d from %d, bps=%d\n",size,fragmentsize,bps);
size=16;
}
#endif
#ifdef OLD_ALSA
buffer_size = size;
memset(&pparams,0,sizeof(pparams));
pparams.fragment_size=size;
pparams.fragments_max=-1; /* choose the best */
pparams.fragments_room=-1;
if(alsa_pcm_playback_params(pcm_h,&pparams)<0)
{
alsa_pcm_close(pcm_h);
pcm_h=NULL;
continue;
}
#else
/* choose all parameters */
err = alsa_pcm_hw_params_any(pcm_h, hwparams);
if (err < 0) {
printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
goto END;
}
{
snd_pcm_uframes_t temp_u_buffer_size, temp_u_period_size;
err = alsa_pcm_get_params(pcm_h, &temp_u_buffer_size, &temp_u_period_size);
if (err < 0) {
alsa_pcm_close(pcm_h);
pcm_h=NULL;
printf("Unable to get buffer size for playback: %s\n", alsa_strerror(err));
goto END;
}
buffer_size_in_frames = 1200;
period_size = temp_u_period_size;
}
/* The set_swparams function was taken from test/pcm.c
* in the alsa-lib distribution*/
if ((err = set_swparams(pcm_h, swparams)) < 0) {
printf("Setting of swparams failed: %s\n", snd_strerror(err));
goto END;
}
#endif
if (!(audiobuffer=(SBYTE*)MikMod_malloc(
#ifdef OLD_ALSA
fragmentsize
#else
buffer_size_in_frames * global_frame_size
#endif
))) {
#ifdef OLD_ALSA
alsa_ctl_close(ctl_h);
#else
alsa_pcm_close(pcm_h);
#endif
return 1;
}
/* sound device is ready to work */
if (VC_Init()) {
#ifdef OLD_ALSA
alsa_ctl_close(ctl_h);
#else
alsa_pcm_close(pcm_h);
#endif
return 1;
} else
return 0;
}
#ifdef OLD_ALSA
alsa_ctl_close(ctl_h);
#else
alsa_pcm_close(pcm_h);
#endif
}
END:
return 1;
}