static void set_params(void)
{
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
snd_pcm_uframes_t buffer_size;
int err;
size_t n;
snd_pcm_uframes_t xfer_align;
unsigned int rate;
snd_pcm_uframes_t start_threshold, stop_threshold;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(handle, params);
if (err < 0) {
error(_("Broken configuration for this PCM: no configurations available"));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
error(_("Access type not available"));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_format(handle, params, hwparams.format);
if (err < 0) {
error(_("Sample format non available"));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels);
if (err < 0) {
error(_("Channels count non available"));
exit(EXIT_FAILURE);
}
#if 0
err = snd_pcm_hw_params_set_periods_min(handle, params, 2);
assert(err >= 0);
#endif
rate = hwparams.rate;
err = snd_pcm_hw_params_set_rate_near(handle, params, &hwparams.rate, 0);
assert(err >= 0);
if ((float)rate * 1.05 < hwparams.rate || (float)rate * 0.95 > hwparams.rate) {
if (!quiet_mode) {
char plugex[64];
const char *pcmname = snd_pcm_name(handle);
fprintf(stderr, _("Warning: rate is not accurate (requested = %iHz, got = %iHz)\n"), rate, hwparams.rate);
if (! pcmname || strchr(snd_pcm_name(handle), ':'))
*plugex = 0;
else
snprintf(plugex, sizeof(plugex), "(-Dplug:%s)",
snd_pcm_name(handle));
fprintf(stderr, _(" please, try the plug plugin %s\n"),
plugex);
}
}
rate = hwparams.rate;
if (buffer_time == 0 && buffer_frames == 0) {
err = snd_pcm_hw_params_get_buffer_time_max(params,
&buffer_time, 0);
assert(err >= 0);
if (buffer_time > 500000)
buffer_time = 500000;
}
if (period_time == 0 && period_frames == 0) {
if (buffer_time > 0)
period_time = buffer_time / 4;
else
period_frames = buffer_frames / 4;
}
if (period_time > 0)
err = snd_pcm_hw_params_set_period_time_near(handle, params,
&period_time, 0);
else
err = snd_pcm_hw_params_set_period_size_near(handle, params,
&period_frames, 0);
assert(err >= 0);
if (buffer_time > 0) {
err = snd_pcm_hw_params_set_buffer_time_near(handle, params,
&buffer_time, 0);
} else {
err = snd_pcm_hw_params_set_buffer_size_near(handle, params,
&buffer_frames);
}
assert(err >= 0);
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
error(_("Unable to install hw params:"));
snd_pcm_hw_params_dump(params, log);
exit(EXIT_FAILURE);
}
snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
if (chunk_size == buffer_size) {
error(_("Can't use period equal to buffer size (%lu == %lu)"),
chunk_size, buffer_size);
exit(EXIT_FAILURE);
}
snd_pcm_sw_params_current(handle, swparams);
err = snd_pcm_sw_params_get_xfer_align(swparams, &xfer_align);
if (err < 0) {
error(_("Unable to obtain xfer align\n"));
exit(EXIT_FAILURE);
}
if (sleep_min)
xfer_align = 1;
err = snd_pcm_sw_params_set_sleep_min(handle, swparams,
sleep_min);
assert(err >= 0);
if (avail_min < 0)
n = chunk_size;
else
n = (double) rate * avail_min / 1000000;
err = snd_pcm_sw_params_set_avail_min(handle, swparams, n);
// round up to closest transfer boundary
n = (buffer_size / xfer_align) * xfer_align;
start_threshold = n;
if (start_threshold < 1)
start_threshold = 1;
if (start_threshold > n)
start_threshold = n;
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
assert(err >= 0);
stop_threshold = buffer_size;
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
assert(err >= 0);
err = snd_pcm_sw_params_set_xfer_align(handle, swparams, xfer_align);
assert(err >= 0);
if (snd_pcm_sw_params(handle, swparams) < 0) {
error(_("unable to install sw params:"));
snd_pcm_sw_params_dump(swparams, log);
exit(EXIT_FAILURE);
}
bits_per_sample = snd_pcm_format_physical_width(hwparams.format);
bits_per_frame = bits_per_sample * hwparams.channels;
chunk_bytes = chunk_size * bits_per_frame / 8;
audiobuf = realloc(audiobuf, chunk_bytes);
if (audiobuf == NULL) {
error(_("not enough memory"));
exit(EXIT_FAILURE);
}
// fprintf(stderr, "real chunk_size = %i, frags = %i, total = %i\n", chunk_size, setup.buf.block.frags, setup.buf.block.frags * chunk_size);
}
int
sound_lowlevel_init( const char *device, int *freqptr, int *stereoptr )
{
unsigned int exact_rate, periods;
unsigned int val, n;
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
snd_pcm_uframes_t avail_min = 0, sound_periodsize, bsize = 0;
static int first_init = 1;
static int init_running = 0;
const char *option;
char tmp;
int err, dir, nperiods = NUM_FRAMES;
float hz;
if( init_running )
return 0;
init_running = 1;
/* select a default device if we weren't explicitly given one */
option = device;
while( option && *option ) {
tmp = '*';
if( ( err = sscanf( option, " buffer=%i %n%c", &val, &n, &tmp ) > 0 ) &&
( tmp == ',' || strlen( option ) == n ) ) {
if( val < 1 ) {
fprintf( stderr, "Bad value for ALSA buffer size %i, using default\n",
val );
} else {
bsize = val;
}
} else if( ( err = sscanf( option, " frames=%i %n%c", &val, &n, &tmp ) > 0 ) &&
( tmp == ',' || strlen( option ) == n ) ) {
if( val < 1 ) {
fprintf( stderr, "Bad value for ALSA buffer size %i frames, using default (%d)\n",
val, NUM_FRAMES );
} else {
nperiods = val;
}
} else if( ( err = sscanf( option, " avail=%i %n%c", &val, &n, &tmp ) > 0 ) &&
( tmp == ',' || strlen( option ) == n ) ) {
if( val < 1 ) {
fprintf( stderr, "Bad value for ALSA avail_min size %i frames, using default\n",
val );
} else {
avail_min = val;
}
} else if( ( err = sscanf( option, " verbose %n%c", &n, &tmp ) == 1 ) &&
( tmp == ',' || strlen( option ) == n ) ) {
verb = 1;
} else { /* try as device name */
while( isspace(*option) )
option++;
if( *option == '\'' ) /* force device... */
option++;
pcm_name = option;
n = strlen( pcm_name );
}
option += n + ( tmp == ',' );
}
/* Open the sound device
*/
if( pcm_name == NULL || *pcm_name == '\0' )
pcm_name = "default";
if( snd_pcm_open( &pcm_handle, pcm_name , stream, 0 ) < 0 ) {
if( strcmp( pcm_name, "default" ) == 0 ) {
/* we try a last one: plughw:0,0 but what a weired ALSA conf.... */
if( snd_pcm_open( &pcm_handle, "plughw:0,0", stream, 0 ) < 0 ) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR,
"couldn't open sound device 'default' and 'plughw:0,0' check ALSA configuration."
);
init_running = 0;
return 1;
} else {
if( first_init )
fprintf( stderr,
"Couldn't open sound device 'default', using 'plughw:0,0' check ALSA configuration.\n"
);
}
}
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR, "couldn't open sound device '%s'.", pcm_name );
init_running = 0;
return 1;
}
/* Allocate the snd_pcm_hw_params_t structure on the stack. */
snd_pcm_hw_params_alloca( &hw_params );
/* Init hw_params with full configuration space */
if( snd_pcm_hw_params_any( pcm_handle, hw_params ) < 0 ) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR,
"couldn't get configuration space on sound device '%s'.",
pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( snd_pcm_hw_params_set_access( pcm_handle, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED ) < 0) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR, "couldn't set access interleaved on '%s'.",
pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
/* Set sample format */
if( snd_pcm_hw_params_set_format( pcm_handle, hw_params,
#if defined WORDS_BIGENDIAN
SND_PCM_FORMAT_S16_BE
#else
SND_PCM_FORMAT_S16_LE
#endif
) < 0 ) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR, "couldn't set format on '%s'.", pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
ch = *stereoptr ? 2 : 1;
if( snd_pcm_hw_params_set_channels( pcm_handle, hw_params, ch )
< 0 ) {
fprintf( stderr, "Couldn't set %s to '%s'.\n", pcm_name,
(*stereoptr ? "stereo" : "mono") );
ch = *stereoptr ? 1 : 2; /* try with opposite */
if( snd_pcm_hw_params_set_channels( pcm_handle, hw_params, ch )
< 0 ) {
ui_error( UI_ERROR_ERROR, "couldn't set %s to '%s'.", pcm_name,
(*stereoptr ? "stereo" : "mono") );
settings_current.sound = 0;
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
*stereoptr = *stereoptr ? 0 : 1; /* write back */
}
framesize = ch << 1; /* we always use 16 bit sorry :-( */
/* Set sample rate. If the exact rate is not supported */
/* by the hardware, use nearest possible rate. */
exact_rate = *freqptr;
if( snd_pcm_hw_params_set_rate_near( pcm_handle, hw_params, &exact_rate,
NULL ) < 0) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR, "couldn't set rate %d on '%s'.",
*freqptr, pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( first_init && *freqptr != exact_rate ) {
fprintf( stderr,
"The rate %d Hz is not supported by your hardware. "
"Using %d Hz instead.\n", *freqptr, exact_rate );
*freqptr = exact_rate;
}
if( bsize != 0 ) {
exact_periodsize = sound_periodsize = bsize / nperiods;
if( bsize < 1 ) {
fprintf( stderr,
"bad value for ALSA buffer size %i, using default.\n",
val );
bsize = 0;
}
}
if( bsize == 0 ) {
/* Adjust relative processor speed to deal with adjusting sound generation
frequency against emulation speed (more flexible than adjusting generated
sample rate) */
hz = (float)sound_get_effective_processor_speed() /
machine_current->timings.tstates_per_frame;
/* Amount of audio data we will accumulate before yielding back to the OS.
Not much point having more than 100Hz playback, we probably get
downgraded by the OS as being a hog too (unlimited Hz limits playback
speed to about 2000% on my Mac, 100Hz allows up to 5000% for me) */
if( hz > 100.0 ) hz = 100.0;
exact_periodsize = sound_periodsize = *freqptr / hz;
}
dir = -1;
if( snd_pcm_hw_params_set_period_size_near( pcm_handle, hw_params,
&exact_periodsize, &dir ) < 0 ) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR, "couldn't set period size %d on '%s'.",
(int)sound_periodsize, pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( first_init && ( exact_periodsize < sound_periodsize / 1.5 ||
exact_periodsize > sound_periodsize * 1.5 ) ) {
fprintf( stderr,
"The period size %d is not supported by your hardware. "
"Using %d instead.\n", (int)sound_periodsize,
(int)exact_periodsize );
}
periods = nperiods;
/* Set number of periods. Periods used to be called fragments. */
if( snd_pcm_hw_params_set_periods_near( pcm_handle, hw_params, &periods,
NULL ) < 0 ) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR, "couldn't set periods on '%s'.", pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( first_init && periods != nperiods ) {
fprintf( stderr, "%d periods is not supported by your hardware. "
"Using %d instead.\n", nperiods, periods );
}
snd_pcm_hw_params_get_buffer_size( hw_params, &exact_bsize );
/* Apply HW parameter settings to */
/* PCM device and prepare device */
if( snd_pcm_hw_params( pcm_handle, hw_params ) < 0 ) {
settings_current.sound = 0;
ui_error( UI_ERROR_ERROR,"couldn't set hw_params on %s", pcm_name );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
snd_pcm_sw_params_alloca( &sw_params );
if( ( err = snd_pcm_sw_params_current( pcm_handle, sw_params ) ) < 0 ) {
ui_error( UI_ERROR_ERROR,"couldn't get sw_params from %s: %s", pcm_name,
snd_strerror ( err ) );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( ( err = snd_pcm_sw_params_set_start_threshold( pcm_handle,
sw_params, exact_periodsize * ( nperiods - 1 ) ) ) < 0 ) {
ui_error( UI_ERROR_ERROR,"couldn't set start_treshold on %s: %s", pcm_name,
snd_strerror ( err ) );
snd_pcm_close( pcm_handle );
init_running = 0;
return 1;
}
if( !avail_min )
avail_min = exact_periodsize >> 1;
if( snd_pcm_sw_params_set_avail_min( pcm_handle,
sw_params, avail_min ) < 0 ) {
#if SND_LIB_VERSION < 0x10010
if( ( err = snd_pcm_sw_params_set_sleep_min( pcm_handle,
sw_params, 1 ) ) < 0 ) {
fprintf( stderr, "Unable to set minimal sleep 1 for %s: %s\n", pcm_name,
snd_strerror ( err ) );
}
#else
fprintf( stderr, "Unable to set avail min %s: %s\n", pcm_name,
snd_strerror( err ) );
#endif
}
#if SND_LIB_VERSION < 0x10010
if( ( err = snd_pcm_sw_params_set_xfer_align( pcm_handle, sw_params, 1 ) ) < 0 ) {
ui_error( UI_ERROR_ERROR,"couldn't set xfer_allign on %s: %s", pcm_name,
snd_strerror ( err ) );
init_running = 0;
return 1;
}
#endif
if( ( err = snd_pcm_sw_params( pcm_handle, sw_params ) ) < 0 ) {
ui_error( UI_ERROR_ERROR,"couldn't set sw_params on %s: %s", pcm_name,
snd_strerror ( err ) );
init_running = 0;
return 1;
}
if( first_init ) snd_output_stdio_attach(&output, stdout, 0);
first_init = 0;
init_running = 0;
return 0; /* success */
}
int ai_alsa_setup(audio_in_t *ai)
{
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
int buffer_size;
int err;
unsigned int rate;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(ai->alsa.handle, params);
if (err < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_PcmBrokenConfig);
return -1;
}
err = snd_pcm_hw_params_set_access(ai->alsa.handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableAccessType);
return -1;
}
err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE);
if (err < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableSampleFmt);
return -1;
}
err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels);
if (err < 0) {
ai->channels = snd_pcm_hw_params_get_channels(params);
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableChanCount,
ai->channels);
} else {
ai->channels = ai->req_channels;
}
err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, ai->req_samplerate, 0);
assert(err >= 0);
rate = err;
ai->samplerate = rate;
ai->alsa.buffer_time = 1000000;
ai->alsa.buffer_time = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params,
ai->alsa.buffer_time, 0);
assert(ai->alsa.buffer_time >= 0);
ai->alsa.period_time = ai->alsa.buffer_time / 4;
ai->alsa.period_time = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params,
ai->alsa.period_time, 0);
assert(ai->alsa.period_time >= 0);
err = snd_pcm_hw_params(ai->alsa.handle, params);
if (err < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_CannotInstallHWParams);
snd_pcm_hw_params_dump(params, ai->alsa.log);
return -1;
}
ai->alsa.chunk_size = snd_pcm_hw_params_get_period_size(params, 0);
buffer_size = snd_pcm_hw_params_get_buffer_size(params);
if (ai->alsa.chunk_size == buffer_size) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_PeriodEqualsBufferSize, ai->alsa.chunk_size, (long)buffer_size);
return -1;
}
snd_pcm_sw_params_current(ai->alsa.handle, swparams);
err = snd_pcm_sw_params_set_sleep_min(ai->alsa.handle, swparams,0);
assert(err >= 0);
err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size);
assert(err >= 0);
err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0);
assert(err >= 0);
err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size);
assert(err >= 0);
assert(err >= 0);
if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_CannotInstallSWParams);
snd_pcm_sw_params_dump(swparams, ai->alsa.log);
return -1;
}
if (mp_msg_test(MSGT_TV, MSGL_V)) {
snd_pcm_dump(ai->alsa.handle, ai->alsa.log);
}
ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels;
ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8;
ai->samplesize = ai->alsa.bits_per_sample;
ai->bytes_per_sample = ai->alsa.bits_per_sample/8;
return 0;
}
int setparams_set(snd_pcm_t *handle,
snd_pcm_hw_params_t *params,
snd_pcm_sw_params_t *swparams,
const char *id)
{
int err;
snd_pcm_uframes_t val;
unsigned int sleep_min = 0;
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
printf("Unable to set hw params for %s: %s\n", id, snd_strerror(err));
return err;
}
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0) {
printf("Unable to determine current swparams for %s: %s\n", id, snd_strerror(err));
return err;
}
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0x7fffffff);
if (err < 0) {
printf("Unable to set start threshold mode for %s: %s\n", id, snd_strerror(err));
return err;
}
tick_time_ok = 0;
if (tick_time > 0) {
int time, ttime;
snd_pcm_hw_params_get_period_time(params, &time, NULL);
snd_pcm_hw_params_get_tick_time(params, &ttime, NULL);
if (time < ttime) {
printf("Skipping to set minimal sleep: period time < tick time\n");
} else if (ttime <= 0) {
printf("Skipping to set minimal sleep: tick time <= 0 (%i)\n", ttime);
} else {
sleep_min = tick_time / ttime;
if (sleep_min <= 0)
sleep_min = 1;
err = snd_pcm_sw_params_set_sleep_min(handle, swparams, sleep_min);
if (err < 0) {
printf("Unable to set minimal sleep %i for %s: %s\n", sleep_min, id, snd_strerror(err));
return err;
}
tick_time_ok = sleep_min * ttime;
}
}
if (!block)
val = 4;
else
snd_pcm_hw_params_get_period_size(params, &val, NULL);
if (tick_time_ok > 0)
val = 16;
err = snd_pcm_sw_params_set_avail_min(handle, swparams, val);
if (err < 0) {
printf("Unable to set avail min for %s: %s\n", id, snd_strerror(err));
return err;
}
val = !block ? 4 : 1;
err = snd_pcm_sw_params_set_xfer_align(handle, swparams, val);
if (err < 0) {
printf("Unable to set transfer align for %s: %s\n", id, snd_strerror(err));
return err;
}
err = snd_pcm_sw_params(handle, swparams);
if (err < 0) {
printf("Unable to set sw params for %s: %s\n", id, snd_strerror(err));
return err;
}
return 0;
}
/*****************************************************************************
* Open: create a handle and open an alsa device
*****************************************************************************
* This function opens an alsa device, through the alsa API.
*
* Note: the only heap-allocated string is psz_device. All the other pointers
* are references to psz_device or to stack-allocated data.
*****************************************************************************/
static int Open( vlc_object_t *p_this )
{
aout_instance_t * p_aout = (aout_instance_t *)p_this;
struct aout_sys_t * p_sys;
vlc_value_t val;
char psz_default_iec_device[128]; /* Buffer used to store the default
S/PDIF device */
char * psz_device, * psz_iec_device; /* device names for PCM and S/PDIF
output */
int i_vlc_pcm_format; /* Audio format for VLC's data */
int i_snd_pcm_format; /* Audio format for ALSA's data */
snd_pcm_uframes_t i_buffer_size = 0;
snd_pcm_uframes_t i_period_size = 0;
int i_channels = 0;
snd_pcm_hw_params_t *p_hw;
snd_pcm_sw_params_t *p_sw;
int i_snd_rc = -1;
unsigned int i_old_rate;
bool b_retry = true;
/* Allocate structures */
p_aout->output.p_sys = p_sys = malloc( sizeof( aout_sys_t ) );
if( p_sys == NULL )
return VLC_ENOMEM;
p_sys->b_playing = false;
p_sys->start_date = 0;
vlc_cond_init( &p_sys->wait );
vlc_mutex_init( &p_sys->lock );
/* Get device name */
if( (psz_device = config_GetPsz( p_aout, "alsa-audio-device" )) == NULL )
{
msg_Err( p_aout, "no audio device given (maybe \"default\" ?)" );
dialog_Fatal( p_aout, _("No Audio Device"), "%s",
_("No audio device name was given. You might want to " \
"enter \"default\".") );
free( p_sys );
return VLC_EGENERIC;
}
/* Choose the IEC device for S/PDIF output:
if the device is overriden by the user then it will be the one
otherwise we compute the default device based on the output format. */
if( AOUT_FMT_NON_LINEAR( &p_aout->output.output )
&& !strcmp( psz_device, DEFAULT_ALSA_DEVICE ) )
{
snprintf( psz_default_iec_device, sizeof(psz_default_iec_device),
"iec958:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x",
IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO,
IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
0,
( p_aout->output.output.i_rate == 48000 ?
IEC958_AES3_CON_FS_48000 :
( p_aout->output.output.i_rate == 44100 ?
IEC958_AES3_CON_FS_44100 : IEC958_AES3_CON_FS_32000 ) ) );
psz_iec_device = psz_default_iec_device;
}
else if( AOUT_FMT_NON_LINEAR( &p_aout->output.output ) )
{
psz_iec_device = psz_device;
}
else
{
psz_iec_device = NULL;
}
/* Choose the linear PCM format (read the comment above about FPU
and float32) */
if( vlc_CPU() & CPU_CAPABILITY_FPU )
{
i_vlc_pcm_format = VLC_CODEC_FL32;
i_snd_pcm_format = SND_PCM_FORMAT_FLOAT;
}
else
{
i_vlc_pcm_format = VLC_CODEC_S16N;
i_snd_pcm_format = SND_PCM_FORMAT_S16;
}
/* If the variable doesn't exist then it's the first time we're called
and we have to probe the available audio formats and channels */
if ( var_Type( p_aout, "audio-device" ) == 0 )
{
Probe( p_aout, psz_device, psz_iec_device, &i_snd_pcm_format );
}
if ( var_Get( p_aout, "audio-device", &val ) < 0 )
{
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
p_aout->output.output.i_format = i_vlc_pcm_format;
if ( val.i_int == AOUT_VAR_5_1 )
{
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT
| AOUT_CHAN_LFE;
free( psz_device );
psz_device = strdup( "surround51" );
}
else if ( val.i_int == AOUT_VAR_2F2R )
{
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT;
free( psz_device );
psz_device = strdup( "surround40" );
}
else if ( val.i_int == AOUT_VAR_STEREO )
{
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
}
else if ( val.i_int == AOUT_VAR_MONO )
{
p_aout->output.output.i_physical_channels = AOUT_CHAN_CENTER;
}
else if( val.i_int != AOUT_VAR_SPDIF )
{
/* This should not happen ! */
msg_Err( p_aout, "internal: can't find audio-device (%i)", val.i_int );
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
#ifdef ALSA_DEBUG
snd_output_stdio_attach( &p_sys->p_snd_stderr, stderr, 0 );
#endif
/* Open the device */
if ( val.i_int == AOUT_VAR_SPDIF )
{
if ( ( i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_iec_device,
SND_PCM_STREAM_PLAYBACK, 0 ) ) < 0 )
{
msg_Err( p_aout, "cannot open ALSA device `%s' (%s)",
psz_iec_device, snd_strerror( i_snd_rc ) );
dialog_Fatal( p_aout, _("Audio output failed"),
_("VLC could not open the ALSA device \"%s\" (%s)."),
psz_iec_device, snd_strerror( i_snd_rc ) );
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
i_buffer_size = ALSA_SPDIF_BUFFER_SIZE;
i_snd_pcm_format = SND_PCM_FORMAT_S16;
i_channels = 2;
i_vlc_pcm_format = VLC_CODEC_SPDIFL;
p_aout->output.i_nb_samples = i_period_size = ALSA_SPDIF_PERIOD_SIZE;
p_aout->output.output.i_bytes_per_frame = AOUT_SPDIF_SIZE;
p_aout->output.output.i_frame_length = A52_FRAME_NB;
aout_VolumeNoneInit( p_aout );
}
else
{
int i;
msg_Dbg( p_aout, "opening ALSA device `%s'", psz_device );
/* Since it seems snd_pcm_close hasn't really released the device at
the time it returns, probe if the device is available in loop for 1s.
We cannot use blocking mode since the we would wait indefinitely when
switching from a dmx device to surround51. */
for( i = 10; i >= 0; i-- )
{
if ( ( i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_device,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) ) == -EBUSY )
{
if( i ) msleep( 100000 /* 100ms */ );
else
{
msg_Err( p_aout, "audio device: %s is already in use",
psz_device );
dialog_Fatal( p_aout, _("Audio output failed"),
_("The audio device \"%s\" is already in use."),
psz_device );
}
continue;
}
break;
}
if( i_snd_rc < 0 )
{
msg_Err( p_aout, "cannot open ALSA device `%s' (%s)",
psz_device, snd_strerror( i_snd_rc ) );
dialog_Fatal( p_aout, _("Audio output failed"),
_("VLC could not open the ALSA device \"%s\" (%s)."),
psz_device, snd_strerror( i_snd_rc ) );
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
/* We want blocking mode */
snd_pcm_nonblock( p_sys->p_snd_pcm, 0 );
i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE;
i_channels = aout_FormatNbChannels( &p_aout->output.output );
p_aout->output.i_nb_samples = i_period_size = ALSA_DEFAULT_PERIOD_SIZE;
aout_VolumeSoftInit( p_aout );
}
/* Free psz_device so that all the remaining data is stack-allocated */
free( psz_device );
p_aout->output.pf_play = Play;
snd_pcm_hw_params_alloca(&p_hw);
snd_pcm_sw_params_alloca(&p_sw);
/* Due to some bugs in alsa with some drivers, we need to retry in s16l
if snd_pcm_hw_params fails in fl32 */
while ( b_retry )
{
b_retry = false;
/* Get Initial hardware parameters */
if ( ( i_snd_rc = snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ) ) < 0 )
{
msg_Err( p_aout, "unable to retrieve initial hardware parameters (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set format. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_format( p_sys->p_snd_pcm, p_hw,
i_snd_pcm_format ) ) < 0 )
{
if( i_snd_pcm_format != SND_PCM_FORMAT_S16 )
{
i_snd_pcm_format = SND_PCM_FORMAT_S16;
i_snd_rc = snd_pcm_hw_params_set_format( p_sys->p_snd_pcm,
p_hw, i_snd_pcm_format );
}
if ( i_snd_rc < 0 )
{
msg_Err( p_aout, "unable to set stream sample size and "
"word order (%s)", snd_strerror( i_snd_rc ) );
goto error;
}
}
if( i_vlc_pcm_format != VLC_CODEC_SPDIFL )
switch( i_snd_pcm_format )
{
case SND_PCM_FORMAT_FLOAT:
i_vlc_pcm_format = VLC_CODEC_FL32;
break;
case SND_PCM_FORMAT_S16:
i_vlc_pcm_format = VLC_CODEC_S16N;
break;
}
p_aout->output.output.i_format = i_vlc_pcm_format;
if ( ( i_snd_rc = snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw,
SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
{
msg_Err( p_aout, "unable to set interleaved stream format (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set channels. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm, p_hw,
i_channels ) ) < 0 )
{
msg_Err( p_aout, "unable to set number of output channels (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set rate. */
i_old_rate = p_aout->output.output.i_rate;
i_snd_rc = snd_pcm_hw_params_set_rate_near( p_sys->p_snd_pcm, p_hw,
&p_aout->output.output.i_rate,
NULL );
if( i_snd_rc < 0 || p_aout->output.output.i_rate != i_old_rate )
{
msg_Warn( p_aout, "The rate %d Hz is not supported by your " \
"hardware. Using %d Hz instead.\n", i_old_rate, \
p_aout->output.output.i_rate );
}
/* Set period size. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_period_size_near( p_sys->p_snd_pcm,
p_hw, &i_period_size, NULL ) ) < 0 )
{
msg_Err( p_aout, "unable to set period size (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
p_aout->output.i_nb_samples = i_period_size;
/* Set buffer size. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm,
p_hw, &i_buffer_size ) ) < 0 )
{
msg_Err( p_aout, "unable to set buffer size (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Commit hardware parameters. */
if ( ( i_snd_rc = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw ) ) < 0 )
{
if ( b_retry == false &&
i_snd_pcm_format == SND_PCM_FORMAT_FLOAT)
{
b_retry = true;
i_snd_pcm_format = SND_PCM_FORMAT_S16;
p_aout->output.output.i_format = VLC_CODEC_S16N;
msg_Warn( p_aout, "unable to commit hardware configuration "
"with fl32 samples. Retrying with s16l (%s)", snd_strerror( i_snd_rc ) );
}
else
{
msg_Err( p_aout, "unable to commit hardware configuration (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
}
}
if( ( i_snd_rc = snd_pcm_hw_params_get_period_time( p_hw,
&p_sys->i_period_time, NULL ) ) < 0 )
{
msg_Err( p_aout, "unable to get period time (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Get Initial software parameters */
snd_pcm_sw_params_current( p_sys->p_snd_pcm, p_sw );
i_snd_rc = snd_pcm_sw_params_set_sleep_min( p_sys->p_snd_pcm, p_sw, 0 );
i_snd_rc = snd_pcm_sw_params_set_avail_min( p_sys->p_snd_pcm, p_sw,
p_aout->output.i_nb_samples );
/* start playing when one period has been written */
i_snd_rc = snd_pcm_sw_params_set_start_threshold( p_sys->p_snd_pcm, p_sw,
ALSA_DEFAULT_PERIOD_SIZE);
if( i_snd_rc < 0 )
{
msg_Err( p_aout, "unable to set start threshold (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Commit software parameters. */
if ( snd_pcm_sw_params( p_sys->p_snd_pcm, p_sw ) < 0 )
{
msg_Err( p_aout, "unable to set software configuration" );
goto error;
}
#ifdef ALSA_DEBUG
snd_output_printf( p_sys->p_snd_stderr, "\nALSA hardware setup:\n\n" );
snd_pcm_dump_hw_setup( p_sys->p_snd_pcm, p_sys->p_snd_stderr );
snd_output_printf( p_sys->p_snd_stderr, "\nALSA software setup:\n\n" );
snd_pcm_dump_sw_setup( p_sys->p_snd_pcm, p_sys->p_snd_stderr );
snd_output_printf( p_sys->p_snd_stderr, "\n" );
#endif
/* Create ALSA thread and wait for its readiness. */
if( vlc_thread_create( p_aout, "aout", ALSAThread,
VLC_THREAD_PRIORITY_OUTPUT ) )
{
msg_Err( p_aout, "cannot create ALSA thread (%m)" );
goto error;
}
return 0;
error:
snd_pcm_close( p_sys->p_snd_pcm );
#ifdef ALSA_DEBUG
snd_output_close( p_sys->p_snd_stderr );
#endif
free( p_sys );
return VLC_EGENERIC;
}
