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vlc/modules/audio_output/alsa.c

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/*****************************************************************************
* alsa.c : alsa plugin for vlc
*****************************************************************************
* Copyright (C) 2000-2001 the VideoLAN team
* $Id$
*
* Authors: Henri Fallon <henri@videolan.org> - Original Author
* Jeffrey Baker <jwbaker@acm.org> - Port to ALSA 1.0 API
* John Paul Lorenti <jpl31@columbia.edu> - Device selection
* Arnaud de Bossoreille de Ribou <bozo@via.ecp.fr> - S/PDIF and aout3
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <assert.h>
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <errno.h> /* ENOMEM */
#include <vlc_dialog.h>
#include <vlc_aout.h>
#include <vlc_cpu.h>
/* ALSA part
Note: we use the new API which is available since 0.9.0beta10a. */
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>
#include <alsa/version.h>
/*#define ALSA_DEBUG*/
/*****************************************************************************
* aout_sys_t: ALSA audio output method descriptor
*****************************************************************************
* This structure is part of the audio output thread descriptor.
* It describes the ALSA specific properties of an audio device.
*****************************************************************************/
struct aout_sys_t
{
snd_pcm_t * p_snd_pcm;
unsigned int i_period_time;
#ifdef ALSA_DEBUG
snd_output_t * p_snd_stderr;
#endif
mtime_t start_date;
vlc_thread_t thread;
vlc_sem_t wait;
};
#define A52_FRAME_NB 1536
/* These values are in frames.
To convert them to a number of bytes you have to multiply them by the
number of channel(s) (eg. 2 for stereo) and the size of a sample (eg.
2 for int16_t). */
#define ALSA_DEFAULT_PERIOD_SIZE 1024
#define ALSA_DEFAULT_BUFFER_SIZE ( ALSA_DEFAULT_PERIOD_SIZE << 8 )
#define ALSA_SPDIF_PERIOD_SIZE A52_FRAME_NB
#define ALSA_SPDIF_BUFFER_SIZE ( ALSA_SPDIF_PERIOD_SIZE << 4 )
/* Why << 4 ? --Meuuh */
/* Why not ? --Bozo */
/* Right. --Meuuh */
#define DEFAULT_ALSA_DEVICE "default"
/*****************************************************************************
* Local prototypes
*****************************************************************************/
static int Open ( vlc_object_t * );
static void Close ( vlc_object_t * );
static void Play ( aout_instance_t * );
static void* ALSAThread ( void * );
static void ALSAFill ( aout_instance_t * );
static int FindDevicesCallback( vlc_object_t *p_this, char const *psz_name,
vlc_value_t newval, vlc_value_t oldval, void *p_unused );
static void GetDevices( vlc_object_t *, module_config_t * );
/*****************************************************************************
* Module descriptor
*****************************************************************************/
static const char *const ppsz_devices[] = {
"default", "plug:front",
"plug:side", "plug:rear", "plug:center_lfe",
"plug:surround40", "plug:surround41",
"plug:surround50", "plug:surround51",
"plug:surround71",
"hdmi", "iec958",
};
static const char *const ppsz_devices_text[] = {
N_("Default"), N_("Front speakers"),
N_("Side speakers"), N_("Rear speakers"), N_("Center and subwoofer"),
N_("Surround 4.0"), N_("Surround 4.1"),
N_("Surround 5.0"), N_("Surround 5.1"),
N_("Surround 7.1"),
N_("HDMI"), N_("S/PDIF"),
};
vlc_module_begin ()
set_shortname( "ALSA" )
set_description( N_("ALSA audio output") )
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_AOUT )
add_string( "alsa-audio-device", DEFAULT_ALSA_DEVICE,
N_("ALSA Device Name"), NULL, false )
add_deprecated_alias( "alsadev" ) /* deprecated since 0.9.3 */
change_string_list( ppsz_devices, ppsz_devices_text, FindDevicesCallback )
change_action_add( FindDevicesCallback, N_("Refresh list") )
set_capability( "audio output", 150 )
set_callbacks( Open, Close )
vlc_module_end ()
/* VLC will insert a resampling filter in any case, so it is best to turn off
* ALSA (plug) resampling. */
static const int mode = SND_PCM_NO_AUTO_RESAMPLE
/* VLC is currently unable to leverage ALSA softvol. Disable it. */
| SND_PCM_NO_SOFTVOL;
/**
* Initializes list of devices.
*/
static void Probe (vlc_object_t *obj)
{
/* Due to design bug in audio output core, this hack is required: */
if (var_Type (obj, "audio-device"))
return;
/* The variable does not exist - first call. */
vlc_value_t text;
var_Create (obj, "audio-device", VLC_VAR_STRING | VLC_VAR_HASCHOICE);
text.psz_string = _("Audio Device");
var_Change (obj, "audio-device", VLC_VAR_SETTEXT, &text, NULL);
GetDevices (obj, NULL);
var_AddCallback (obj, "audio-device", aout_ChannelsRestart, NULL);
var_TriggerCallback (obj, "intf-change");
}
/*****************************************************************************
* 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 *obj)
{
aout_instance_t * p_aout = (aout_instance_t *)obj;
/* Get device name */
char *psz_device;
if (var_Type (p_aout, "audio-device"))
psz_device = var_GetString (p_aout, "audio-device");
else
psz_device = var_InheritString( p_aout, "alsa-audio-device" );
if (unlikely(psz_device == NULL))
return VLC_ENOMEM;
snd_pcm_format_t pcm_format; /* ALSA sample format */
vlc_fourcc_t fourcc = p_aout->output.output.i_format;
bool spdif = false;
switch (fourcc)
{
case VLC_CODEC_F64B:
pcm_format = SND_PCM_FORMAT_FLOAT64_BE;
break;
case VLC_CODEC_F64L:
pcm_format = SND_PCM_FORMAT_FLOAT64_LE;
break;
case VLC_CODEC_F32B:
pcm_format = SND_PCM_FORMAT_FLOAT_BE;
break;
case VLC_CODEC_F32L:
pcm_format = SND_PCM_FORMAT_FLOAT_LE;
break;
case VLC_CODEC_FI32:
fourcc = VLC_CODEC_FL32;
pcm_format = SND_PCM_FORMAT_FLOAT;
break;
case VLC_CODEC_S32B:
pcm_format = SND_PCM_FORMAT_S32_BE;
break;
case VLC_CODEC_S32L:
pcm_format = SND_PCM_FORMAT_S32_LE;
break;
case VLC_CODEC_S24B:
pcm_format = SND_PCM_FORMAT_S24_3BE;
break;
case VLC_CODEC_S24L:
pcm_format = SND_PCM_FORMAT_S24_3LE;
break;
case VLC_CODEC_U24B:
pcm_format = SND_PCM_FORMAT_U24_3BE;
break;
case VLC_CODEC_U24L:
pcm_format = SND_PCM_FORMAT_U24_3LE;
break;
case VLC_CODEC_S16B:
pcm_format = SND_PCM_FORMAT_S16_BE;
break;
case VLC_CODEC_S16L:
pcm_format = SND_PCM_FORMAT_S16_LE;
break;
case VLC_CODEC_U16B:
pcm_format = SND_PCM_FORMAT_U16_BE;
break;
case VLC_CODEC_U16L:
pcm_format = SND_PCM_FORMAT_U16_LE;
break;
case VLC_CODEC_S8:
pcm_format = SND_PCM_FORMAT_S8;
break;
case VLC_CODEC_U8:
pcm_format = SND_PCM_FORMAT_U8;
break;
default:
if (AOUT_FMT_NON_LINEAR(&p_aout->output.output))
spdif = var_InheritBool (p_aout, "spdif");
if (HAVE_FPU)
{
fourcc = VLC_CODEC_FL32;
pcm_format = SND_PCM_FORMAT_FLOAT;
}
else
{
fourcc = VLC_CODEC_S16N;
pcm_format = SND_PCM_FORMAT_S16;
}
}
/* Choose the IEC device for S/PDIF output:
if the device is overridden by the user then it will be the one
otherwise we compute the default device based on the output format. */
if (spdif && !strcmp (psz_device, DEFAULT_ALSA_DEVICE))
{
unsigned aes3;
switch (p_aout->output.output.i_rate)
{
#define FS(freq) \
case freq: aes3 = IEC958_AES3_CON_FS_ ## freq; break;
FS( 44100) /* def. */ FS( 48000) FS( 32000)
FS( 22050) FS( 24000)
FS( 88200) FS(768000) FS( 96000)
FS(176400) FS(192000)
#undef FS
default:
aes3 = IEC958_AES3_CON_FS_NOTID;
break;
}
free (psz_device);
if (asprintf (&psz_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, aes3) == -1)
return VLC_ENOMEM;
}
/* Allocate structures */
aout_sys_t *p_sys = malloc (sizeof (*p_sys));
if (unlikely(p_sys == NULL))
{
free (psz_device);
return VLC_ENOMEM;
}
p_aout->output.p_sys = p_sys;
#ifdef ALSA_DEBUG
snd_output_stdio_attach( &p_sys->p_snd_stderr, stderr, 0 );
#endif
/* Open the device */
msg_Dbg( p_aout, "opening ALSA device `%s'", psz_device );
int val = snd_pcm_open (&p_sys->p_snd_pcm, psz_device,
SND_PCM_STREAM_PLAYBACK, mode);
#if (SND_LIB_VERSION <= 0x010015)
# warning Please update alsa-lib to version > 1.0.21a.
var_Create (p_aout->p_libvlc, "alsa-working", VLC_VAR_BOOL);
if (val != 0 && var_GetBool (p_aout->p_libvlc, "alsa-working"))
dialog_Fatal (p_aout, "ALSA version problem",
"VLC failed to re-initialize your audio output device.\n"
"Please update alsa-lib to version 1.0.22 or higher "
"to fix this issue.");
var_SetBool (p_aout->p_libvlc, "alsa-working", !val);
#endif
if (val != 0)
{
#if (SND_LIB_VERSION <= 0x010017)
# warning Please update alsa-lib to version > 1.0.23.
var_Create (p_aout->p_libvlc, "alsa-broken", VLC_VAR_BOOL);
if (!var_GetBool (p_aout->p_libvlc, "alsa-broken"))
{
var_SetBool (p_aout->p_libvlc, "alsa-broken", true);
dialog_Fatal (p_aout, "Potential ALSA version problem",
"VLC failed to initialize your audio output device (if any).\n"
"Please update alsa-lib to version 1.0.24 or higher "
"to try to fix this issue.");
}
#endif
msg_Err (p_aout, "cannot open ALSA device `%s' (%s)",
psz_device, snd_strerror (val));
dialog_Fatal (p_aout, _("Audio output failed"),
_("The audio device \"%s\" could not be used:\n%s."),
psz_device, snd_strerror (val));
free (psz_device);
free (p_sys);
return VLC_EGENERIC;
}
free( psz_device );
snd_pcm_uframes_t i_buffer_size;
snd_pcm_uframes_t i_period_size;
int i_channels;
if (spdif)
{
fourcc = VLC_CODEC_SPDIFL;
i_buffer_size = ALSA_SPDIF_BUFFER_SIZE;
pcm_format = SND_PCM_FORMAT_S16;
i_channels = 2;
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
{
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 );
}
p_aout->output.pf_play = Play;
snd_pcm_hw_params_t *p_hw;
snd_pcm_sw_params_t *p_sw;
snd_pcm_hw_params_alloca(&p_hw);
snd_pcm_sw_params_alloca(&p_sw);
/* Get Initial hardware parameters */
val = snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw );
if( val < 0 )
{
msg_Err( p_aout, "unable to retrieve hardware parameters (%s)",
snd_strerror( val ) );
goto error;
}
/* Set format. */
val = snd_pcm_hw_params_set_format (p_sys->p_snd_pcm, p_hw, pcm_format);
if( val < 0 )
{
msg_Err (p_aout, "cannot set sample format: %s", snd_strerror (val));
goto error;
}
p_aout->output.output.i_format = fourcc;
val = snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw,
SND_PCM_ACCESS_RW_INTERLEAVED );
if( val < 0 )
{
msg_Err( p_aout, "unable to set interleaved stream format (%s)",
snd_strerror( val ) );
goto error;
}
/* Set channels. */
val = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm, p_hw, i_channels );
if( val < 0 )
{
msg_Err( p_aout, "unable to set number of output channels (%s)",
snd_strerror( val ) );
goto error;
}
/* Set rate. */
unsigned old_rate = p_aout->output.output.i_rate;
val = snd_pcm_hw_params_set_rate_near (p_sys->p_snd_pcm, p_hw,
&p_aout->output.output.i_rate,
NULL);
if (val < 0)
{
msg_Err (p_aout, "unable to set sampling rate (%s)",
snd_strerror (val));
goto error;
}
if (p_aout->output.output.i_rate != old_rate)
msg_Warn (p_aout, "resampling from %d Hz to %d Hz", old_rate,
p_aout->output.output.i_rate);
/* Set period size. */
val = snd_pcm_hw_params_set_period_size_near( p_sys->p_snd_pcm, p_hw,
&i_period_size, NULL );
if( val < 0 )
{
msg_Err( p_aout, "unable to set period size (%s)",
snd_strerror( val ) );
goto error;
}
p_aout->output.i_nb_samples = i_period_size;
/* Set buffer size. */
val = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm, p_hw,
&i_buffer_size );
if( val )
{
msg_Err( p_aout, "unable to set buffer size (%s)",
snd_strerror( val ) );
goto error;
}
/* Commit hardware parameters. */
val = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw );
if( val < 0 )
{
msg_Err( p_aout, "unable to commit hardware configuration (%s)",
snd_strerror( val ) );
goto error;
}
val = snd_pcm_hw_params_get_period_time( p_hw, &p_sys->i_period_time,
NULL );
if( val < 0 )
{
msg_Err( p_aout, "unable to get period time (%s)",
snd_strerror( val ) );
goto error;
}
/* Get Initial software parameters */
snd_pcm_sw_params_current( p_sys->p_snd_pcm, p_sw );
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 */
val = snd_pcm_sw_params_set_start_threshold( p_sys->p_snd_pcm, p_sw,
ALSA_DEFAULT_PERIOD_SIZE);
if( val < 0 )
{
msg_Err( p_aout, "unable to set start threshold (%s)",
snd_strerror( val ) );
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
p_sys->start_date = 0;
vlc_sem_init( &p_sys->wait, 0 );
/* Create ALSA thread and wait for its readiness. */
if( vlc_clone( &p_sys->thread, ALSAThread, p_aout,
VLC_THREAD_PRIORITY_OUTPUT ) )
{
msg_Err( p_aout, "cannot create ALSA thread (%m)" );
vlc_sem_destroy( &p_sys->wait );
goto error;
}
Probe (obj);
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;
}
static void PlayIgnore( aout_instance_t *p_aout )
{ /* Already playing - nothing to do */
(void) p_aout;
}
/*****************************************************************************
* Play: start playback
*****************************************************************************/
static void Play( aout_instance_t *p_aout )
{
p_aout->output.pf_play = PlayIgnore;
/* get the playing date of the first aout buffer */
p_aout->output.p_sys->start_date = aout_FifoFirstDate( &p_aout->output.fifo );
/* wake up the audio output thread */
sem_post( &p_aout->output.p_sys->wait );
}
/*****************************************************************************
* Close: close the ALSA device
*****************************************************************************/
static void Close (vlc_object_t *obj)
{
aout_instance_t *p_aout = (aout_instance_t *)obj;
struct aout_sys_t * p_sys = p_aout->output.p_sys;
/* Make sure that the thread will stop once it is waken up */
vlc_cancel( p_sys->thread );
vlc_join( p_sys->thread, NULL );
vlc_sem_destroy( &p_sys->wait );
snd_pcm_drop( p_sys->p_snd_pcm );
snd_pcm_close( p_sys->p_snd_pcm );
#ifdef ALSA_DEBUG
snd_output_close( p_sys->p_snd_stderr );
#endif
free( p_sys );
}
/*****************************************************************************
* ALSAThread: asynchronous thread used to DMA the data to the device
*****************************************************************************/
static void* ALSAThread( void *data )
{
aout_instance_t * p_aout = data;
struct aout_sys_t * p_sys = p_aout->output.p_sys;
/* Wait for the exact time to start playing (avoids resampling) */
vlc_sem_wait( &p_sys->wait );
mwait( p_sys->start_date - AOUT_PTS_TOLERANCE / 4 );
for(;;)
ALSAFill( p_aout );
assert(0);
}
/*****************************************************************************
* ALSAFill: function used to fill the ALSA buffer as much as possible
*****************************************************************************/
static void ALSAFill( aout_instance_t * p_aout )
{
struct aout_sys_t * p_sys = p_aout->output.p_sys;
snd_pcm_t *p_pcm = p_sys->p_snd_pcm;
snd_pcm_status_t * p_status;
int i_snd_rc;
mtime_t next_date;
int canc = vlc_savecancel();
/* Fill in the buffer until space or audio output buffer shortage */
/* Get the status */
snd_pcm_status_alloca(&p_status);
i_snd_rc = snd_pcm_status( p_pcm, p_status );
if( i_snd_rc < 0 )
{
msg_Err( p_aout, "cannot get device status" );
goto error;
}
/* Handle buffer underruns and get the status again */
if( snd_pcm_status_get_state( p_status ) == SND_PCM_STATE_XRUN )
{
/* Prepare the device */
i_snd_rc = snd_pcm_prepare( p_pcm );
if( i_snd_rc )
{
msg_Err( p_aout, "cannot recover from buffer underrun" );
goto error;
}
msg_Dbg( p_aout, "recovered from buffer underrun" );
/* Get the new status */
i_snd_rc = snd_pcm_status( p_pcm, p_status );
if( i_snd_rc < 0 )
{
msg_Err( p_aout, "cannot get device status after recovery" );
goto error;
}
/* Underrun, try to recover as quickly as possible */
next_date = mdate();
}
else
{
/* Here the device should be in RUNNING state, p_status is valid. */
snd_pcm_sframes_t delay = snd_pcm_status_get_delay( p_status );
if( delay == 0 ) /* workaround buggy alsa drivers */
if( snd_pcm_delay( p_pcm, &delay ) < 0 )
delay = 0; /* FIXME: use a positive minimal delay */
size_t i_bytes = snd_pcm_frames_to_bytes( p_pcm, delay );
mtime_t delay_us = CLOCK_FREQ * i_bytes
/ p_aout->output.output.i_bytes_per_frame
/ p_aout->output.output.i_rate
* p_aout->output.output.i_frame_length;
#ifdef ALSA_DEBUG
snd_pcm_state_t state = snd_pcm_status_get_state( p_status );
if( state != SND_PCM_STATE_RUNNING )
msg_Err( p_aout, "pcm status (%d) != RUNNING", state );
msg_Dbg( p_aout, "Delay is %ld frames (%zu bytes)", delay, i_bytes );
msg_Dbg( p_aout, "Bytes per frame: %d", p_aout->output.output.i_bytes_per_frame );
msg_Dbg( p_aout, "Rate: %d", p_aout->output.output.i_rate );
msg_Dbg( p_aout, "Frame length: %d", p_aout->output.output.i_frame_length );
msg_Dbg( p_aout, "Next date: in %"PRId64" microseconds", delay_us );
#endif
next_date = mdate() + delay_us;
}
block_t *p_buffer = aout_OutputNextBuffer( p_aout, next_date,
(p_aout->output.output.i_format == VLC_CODEC_SPDIFL) );
/* Audio output buffer shortage -> stop the fill process and wait */
if( p_buffer == NULL )
goto error;
block_cleanup_push( p_buffer );
for (;;)
{
int n = snd_pcm_poll_descriptors_count(p_pcm);
struct pollfd ufd[n];
unsigned short revents;
snd_pcm_poll_descriptors(p_pcm, ufd, n);
do
{
vlc_restorecancel(canc);
poll(ufd, n, -1);
canc = vlc_savecancel();
snd_pcm_poll_descriptors_revents(p_pcm, ufd, n, &revents);
}
while(!revents);
if(revents & POLLOUT)
{
i_snd_rc = snd_pcm_writei( p_pcm, p_buffer->p_buffer,
p_buffer->i_nb_samples );
if( i_snd_rc != -ESTRPIPE )
break;
}
/* a suspend event occurred
* (stream is suspended and waiting for an application recovery) */
msg_Dbg( p_aout, "entering in suspend mode, trying to resume..." );
while( ( i_snd_rc = snd_pcm_resume( p_pcm ) ) == -EAGAIN )
{
vlc_restorecancel(canc);
msleep(CLOCK_FREQ); /* device still suspended, wait... */
canc = vlc_savecancel();
}
if( i_snd_rc < 0 )
/* Device does not support resuming, restart it */
i_snd_rc = snd_pcm_prepare( p_pcm );
}
if( i_snd_rc < 0 )
msg_Err( p_aout, "cannot write: %s", snd_strerror( i_snd_rc ) );
vlc_restorecancel(canc);
vlc_cleanup_run();
return;
error:
if( i_snd_rc < 0 )
msg_Err( p_aout, "ALSA error: %s", snd_strerror( i_snd_rc ) );
vlc_restorecancel(canc);
msleep(p_sys->i_period_time / 2);
}
/*****************************************************************************
* config variable callback
*****************************************************************************/
static int FindDevicesCallback( vlc_object_t *p_this, char const *psz_name,
vlc_value_t newval, vlc_value_t oldval, void *p_unused )
{
module_config_t *p_item;
(void)newval;
(void)oldval;
(void)p_unused;
p_item = config_FindConfig( p_this, psz_name );
if( !p_item ) return VLC_SUCCESS;
/* Clear-up the current list */
if( p_item->i_list )
{
int i;
/* Keep the first entrie */
for( i = 1; i < p_item->i_list; i++ )
{
free( (char *)p_item->ppsz_list[i] );
free( (char *)p_item->ppsz_list_text[i] );
}
/* TODO: Remove when no more needed */
p_item->ppsz_list[i] = NULL;
p_item->ppsz_list_text[i] = NULL;
}
p_item->i_list = 1;
GetDevices( p_this, p_item );
/* Signal change to the interface */
p_item->b_dirty = true;
return VLC_SUCCESS;
}
static void GetDevices (vlc_object_t *obj, module_config_t *item)
{
void **hints;
msg_Dbg(obj, "Available ALSA PCM devices:");
if (snd_device_name_hint(-1, "pcm", &hints) < 0)
return;
for (size_t i = 0; hints[i] != NULL; i++)
{
void *hint = hints[i];
char *dev;
char *name = snd_device_name_get_hint(hint, "NAME");
if (unlikely(name == NULL))
continue;
if (unlikely(asprintf (&dev, "plug:'%s'", name) == -1))
{
free(name);
continue;
}
char *desc = snd_device_name_get_hint(hint, "DESC");
if (desc != NULL)
for (char *lf = strchr(desc, '\n'); lf; lf = strchr(lf, '\n'))
*lf = ' ';
msg_Dbg(obj, "%s (%s)", (desc != NULL) ? desc : name, name);
if (item != NULL)
{
item->ppsz_list = xrealloc(item->ppsz_list,
(item->i_list + 2) * sizeof(char *));
item->ppsz_list_text = xrealloc(item->ppsz_list_text,
(item->i_list + 2) * sizeof(char *));
item->ppsz_list[item->i_list] = dev;
if (desc == NULL)
desc = strdup(name);
item->ppsz_list_text[item->i_list] = desc;
item->i_list++;
}
else
{
vlc_value_t val, text;
val.psz_string = dev;
text.psz_string = desc;
var_Change(obj, "audio-device", VLC_VAR_ADDCHOICE, &val, &text);
free(desc);
free(dev);
free(name);
}
}
snd_device_name_free_hint(hints);
if (item != NULL)
{
item->ppsz_list[item->i_list] = NULL;
item->ppsz_list_text[item->i_list] = NULL;
}
}