/*****************************************************************************
* alsa.c : alsa plugin for vlc
*****************************************************************************
- * Copyright (C) 2000-2001 the VideoLAN team
- * $Id$
+ * Copyright (C) 2000-2010 VLC authors and VideoLAN
+ * Copyright (C) 2009-2011 RĂ©mi Denis-Courmont
*
* 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
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 2.1 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.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser 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.
+ * You should have received a copy of the GNU Lesser 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 <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.
- *****************************************************************************/
+/** Private data for an ALSA PCM playback stream */
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;
+ snd_pcm_t *pcm;
+ unsigned rate; /**< Sample rate */
+ uint8_t chans_table[AOUT_CHAN_MAX]; /**< Channels order table */
+ uint8_t chans_to_reorder; /**< Number of channels to reorder */
+ uint8_t bits; /**< Bits per sample per channel */
+
+ bool soft_mute;
+ float soft_gain;
+ char *device;
};
-#define A52_FRAME_NB 1536
+#include "volume.h"
-/* 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 A52_FRAME_NB 1536
-#define DEFAULT_ALSA_DEVICE "default"
+static int Open (vlc_object_t *);
+static void Close (vlc_object_t *);
+static int EnumDevices (vlc_object_t *, char const *, char ***, char ***);
-/*****************************************************************************
- * Local prototypes
- *****************************************************************************/
-static int Open ( vlc_object_t * );
-static void Close ( vlc_object_t * );
-static void Play ( audio_output_t * );
-static void* ALSAThread ( void * );
-static void ALSAFill ( audio_output_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 * );
+#define AUDIO_DEV_TEXT N_("Audio output device")
+#define AUDIO_DEV_LONGTEXT N_("Audio output device (using ALSA syntax).")
-/*****************************************************************************
- * 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",
+#define AUDIO_CHAN_TEXT N_("Audio output channels")
+#define AUDIO_CHAN_LONGTEXT N_("Channels available for audio output. " \
+ "If the input has more channels than the output, it will be down-mixed. " \
+ "This parameter is ignored when digital pass-through is active.")
+static const int channels[] = {
+ AOUT_CHAN_CENTER, AOUT_CHANS_STEREO, AOUT_CHANS_4_0, AOUT_CHANS_4_1,
+ AOUT_CHANS_5_0, AOUT_CHANS_5_1, AOUT_CHANS_7_1,
};
-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"),
+static const char *const channels_text[] = {
+ N_("Mono"), N_("Stereo"), N_("Surround 4.0"), N_("Surround 4.1"),
+ N_("Surround 5.0"), N_("Surround 5.1"), N_("Surround 7.1"),
};
+
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") )
-
+ add_string ("alsa-audio-device", "default",
+ AUDIO_DEV_TEXT, AUDIO_DEV_LONGTEXT, false)
+ change_string_cb (EnumDevices)
+ add_integer ("alsa-audio-channels", AOUT_CHANS_FRONT,
+ AUDIO_CHAN_TEXT, AUDIO_CHAN_LONGTEXT, false)
+ change_integer_list (channels, channels_text)
+ add_sw_gain ()
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)
+/** Helper for ALSA -> VLC debugging output */
+static void Dump (vlc_object_t *obj, const char *msg,
+ int (*cb)(void *, snd_output_t *), void *p)
{
- /* Due to design bug in audio output core, this hack is required: */
- if (var_Type (obj, "audio-device"))
+ snd_output_t *output;
+ char *str;
+
+ if (unlikely(snd_output_buffer_open (&output)))
return;
- /* The variable does not exist - first call. */
- vlc_value_t text;
+ int val = cb (p, output);
+ if (val)
+ {
+ msg_Warn (obj, "cannot get info: %s", snd_strerror (val));
+ return;
+ }
+
+ size_t len = snd_output_buffer_string (output, &str);
+ if (len > 0 && str[len - 1])
+ len--; /* strip trailing newline */
+ msg_Dbg (obj, "%s%.*s", msg, (int)len, str);
+ snd_output_close (output);
+}
+#define Dump(o, m, cb, p) \
+ Dump(VLC_OBJECT(o), m, (int (*)(void *, snd_output_t *))(cb), p)
+
+static void DumpDevice (vlc_object_t *obj, snd_pcm_t *pcm)
+{
+ snd_pcm_info_t *info;
- 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);
+ Dump (obj, " ", snd_pcm_dump, pcm);
+ snd_pcm_info_alloca (&info);
+ if (snd_pcm_info (pcm, info) == 0)
+ {
+ msg_Dbg (obj, " device name : %s", snd_pcm_info_get_name (info));
+ msg_Dbg (obj, " device ID : %s", snd_pcm_info_get_id (info));
+ msg_Dbg (obj, " subdevice name: %s",
+ snd_pcm_info_get_subdevice_name (info));
+ }
+}
- GetDevices (obj, NULL);
+static void DumpDeviceStatus (vlc_object_t *obj, snd_pcm_t *pcm)
+{
+ snd_pcm_status_t *status;
- var_AddCallback (obj, "audio-device", aout_ChannelsRestart, NULL);
- var_TriggerCallback (obj, "intf-change");
+ snd_pcm_status_alloca (&status);
+ snd_pcm_status (pcm, status);
+ Dump (obj, "current status:\n", snd_pcm_status_dump, status);
}
+#define DumpDeviceStatus(o, p) DumpDeviceStatus(VLC_OBJECT(o), p)
-/*****************************************************************************
- * 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)
+static unsigned SetupChannelsUnknown (vlc_object_t *obj,
+ uint16_t *restrict mask)
{
- audio_output_t * p_aout = (audio_output_t *)obj;
+ uint16_t map = var_InheritInteger (obj, "alsa-audio-channels");
+ uint16_t chans = *mask & map;
- /* Get device name */
- char *psz_device;
+ if (unlikely(chans == 0)) /* WTH? */
+ chans = AOUT_CHANS_STEREO;
- if (var_Type (p_aout, "audio-device"))
- psz_device = var_GetString (p_aout, "audio-device");
+ if (popcount (chans) < popcount (*mask))
+ msg_Dbg (obj, "downmixing from %u to %u channels",
+ popcount (*mask), popcount (chans));
else
- psz_device = var_InheritString( p_aout, "alsa-audio-device" );
- if (unlikely(psz_device == NULL))
- return VLC_ENOMEM;
+ msg_Dbg (obj, "keeping %u channels", popcount (chans));
+ *mask = chans;
+ return 0;
+}
+
+#if (SND_LIB_VERSION >= 0x01001B)
+static const uint16_t vlc_chans[] = {
+ [SND_CHMAP_MONO] = AOUT_CHAN_CENTER,
+ [SND_CHMAP_FL] = AOUT_CHAN_LEFT,
+ [SND_CHMAP_FR] = AOUT_CHAN_RIGHT,
+ [SND_CHMAP_RL] = AOUT_CHAN_REARLEFT,
+ [SND_CHMAP_RR] = AOUT_CHAN_REARRIGHT,
+ [SND_CHMAP_FC] = AOUT_CHAN_CENTER,
+ [SND_CHMAP_LFE] = AOUT_CHAN_LFE,
+ [SND_CHMAP_SL] = AOUT_CHAN_MIDDLELEFT,
+ [SND_CHMAP_SR] = AOUT_CHAN_MIDDLERIGHT,
+ [SND_CHMAP_RC] = AOUT_CHAN_REARCENTER,
+};
+static int Map2Mask (vlc_object_t *obj, const snd_pcm_chmap_t *restrict map)
+{
+ uint16_t mask = 0;
+
+ for (unsigned i = 0; i < map->channels; i++)
+ {
+ const unsigned pos = map->pos[i];
+ uint_fast16_t vlc_chan = 0;
+
+ if (pos < sizeof (vlc_chans) / sizeof (vlc_chans[0]))
+ vlc_chan = vlc_chans[pos];
+ if (vlc_chan == 0)
+ {
+ msg_Dbg (obj, " %s channel %u position %u", "unsupported", i, pos);
+ return -1;
+ }
+ if (mask & vlc_chan)
+ {
+ msg_Dbg (obj, " %s channel %u position %u", "duplicate", i, pos);
+ return -1;
+ }
+ mask |= vlc_chan;
+ }
+ return mask;
+}
+
+/**
+ * Compares a fixed ALSA channels map with the VLC channels order.
+ */
+static unsigned SetupChannelsFixed(const snd_pcm_chmap_t *restrict map,
+ uint16_t *restrict mask, uint8_t *restrict tab)
+{
+ uint32_t chans_out[AOUT_CHAN_MAX];
+
+ for (unsigned i = 0; i < map->channels; i++)
+ {
+ uint_fast16_t vlc_chan = vlc_chans[map->pos[i]];
+
+ chans_out[i] = vlc_chan;
+ *mask |= vlc_chan;
+ }
+
+ return aout_CheckChannelReorder(NULL, chans_out, *mask, tab);
+}
+
+/**
+ * Negotiate channels mapping.
+ */
+static unsigned SetupChannels (vlc_object_t *obj, snd_pcm_t *pcm,
+ uint16_t *restrict mask, uint8_t *restrict tab)
+{
+ snd_pcm_chmap_query_t **maps = snd_pcm_query_chmaps (pcm);
+ if (tab == NULL)
+ { /* Fallback to manual configuration */
+ msg_Dbg(obj, "channels map not provided");
+ return SetupChannelsUnknown (obj, mask);
+ }
+
+ /* Find most appropriate available channels map */
+ unsigned best_offset;
+ unsigned best_score = 0;
+
+ for (snd_pcm_chmap_query_t *const *p = maps; *p != NULL; p++)
+ {
+ snd_pcm_chmap_query_t *map = *p;
+
+ switch (map->type)
+ {
+ case SND_CHMAP_TYPE_FIXED:
+ case SND_CHMAP_TYPE_PAIRED:
+ case SND_CHMAP_TYPE_VAR:
+ break;
+ default:
+ msg_Err (obj, "unknown channels map type %u", map->type);
+ continue;
+ }
+
+ int chans = Map2Mask (obj, &map->map);
+ if (chans == -1)
+ continue;
+
+ unsigned score = popcount (chans & *mask);
+ if (score > best_score)
+ {
+ best_offset = p - maps;
+ best_score = score;
+ }
+ }
+
+ if (best_score == 0)
+ {
+ msg_Err (obj, "cannot find supported channels map");
+ snd_pcm_free_chmaps (maps);
+ return SetupChannelsUnknown (obj, mask);
+ }
+
+ const snd_pcm_chmap_t *map = &maps[best_offset]->map;
+ msg_Dbg (obj, "using channels map %u, type %u, %u channel(s)", best_offset,
+ maps[best_offset]->type, best_score);
+
+ /* Setup channels map */
+ unsigned to_reorder = SetupChannelsFixed(map, mask, tab);
+
+ /* TODO: avoid reordering for PAIRED and VAR types */
+ //snd_pcm_set_chmap (pcm, ...)
+
+ snd_pcm_free_chmaps (maps);
+ return to_reorder;
+}
+#else /* (SND_LIB_VERSION < 0x01001B) */
+# define SetupChannels(obj, pcm, mask, tab) \
+ SetupChannelsUnknown(obj, mask)
+#endif
+
+static int TimeGet (audio_output_t *aout, mtime_t *);
+static void Play (audio_output_t *, block_t *);
+static void Pause (audio_output_t *, bool, mtime_t);
+static void PauseDummy (audio_output_t *, bool, mtime_t);
+static void Flush (audio_output_t *, bool);
+
+/** Initializes an ALSA playback stream */
+static int Start (audio_output_t *aout, audio_sample_format_t *restrict fmt)
+{
+ aout_sys_t *sys = aout->sys;
snd_pcm_format_t pcm_format; /* ALSA sample format */
- vlc_fourcc_t fourcc = p_aout->format.i_format;
bool spdif = false;
- switch (fourcc)
+ switch (fmt->i_format)
{
- case VLC_CODEC_F64B:
- pcm_format = SND_PCM_FORMAT_FLOAT64_BE;
+ case VLC_CODEC_FL64:
+ pcm_format = SND_PCM_FORMAT_FLOAT64;
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;
+ case 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;
+ case VLC_CODEC_S32N:
+ pcm_format = SND_PCM_FORMAT_S32;
break;
case VLC_CODEC_S24B:
pcm_format = SND_PCM_FORMAT_S24_3BE;
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;
+ case VLC_CODEC_S16N:
+ pcm_format = SND_PCM_FORMAT_S16;
break;
case VLC_CODEC_U8:
pcm_format = SND_PCM_FORMAT_U8;
break;
default:
- if (AOUT_FMT_NON_LINEAR(&p_aout->format))
- spdif = var_InheritBool (p_aout, "spdif");
+ if (AOUT_FMT_SPDIF(fmt))
+ spdif = var_InheritBool (aout, "spdif");
+ if (spdif)
+ {
+ fmt->i_format = VLC_CODEC_SPDIFL;
+ pcm_format = SND_PCM_FORMAT_S16;
+ }
+ else
if (HAVE_FPU)
{
- fourcc = VLC_CODEC_FL32;
+ fmt->i_format = VLC_CODEC_FL32;
pcm_format = SND_PCM_FORMAT_FLOAT;
}
else
{
- fourcc = VLC_CODEC_S16N;
+ fmt->i_format = 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))
+ const char *device = sys->device;
+ char *devbuf = NULL;
+ /* Choose the IEC device for S/PDIF output */
+ if (spdif && !strcmp (device, "default"))
{
unsigned aes3;
- switch (p_aout->format.i_rate)
+ switch (fmt->i_rate)
{
#define FS(freq) \
case freq: aes3 = IEC958_AES3_CON_FS_ ## freq; break;
break;
}
- free (psz_device);
- if (asprintf (&psz_device,
+ if (asprintf (&devbuf,
"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;
+ device = devbuf;
}
- /* Allocate structures */
- aout_sys_t *p_sys = malloc (sizeof (*p_sys));
- if (unlikely(p_sys == NULL))
- {
- free (psz_device);
- return VLC_ENOMEM;
- }
- p_aout->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
+ snd_pcm_t *pcm;
+ /* VLC always has a resampler. No need for ALSA's. */
+ const int mode = SND_PCM_NO_AUTO_RESAMPLE;
+
+ int val = snd_pcm_open (&pcm, device, SND_PCM_STREAM_PLAYBACK, mode);
+ free (devbuf);
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"),
+ msg_Err (aout, "cannot open ALSA device \"%s\": %s", sys->device,
+ snd_strerror (val));
+ dialog_Fatal (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);
+ sys->device, snd_strerror (val));
return VLC_EGENERIC;
}
- free( psz_device );
+ sys->pcm = pcm;
- snd_pcm_uframes_t i_buffer_size;
- snd_pcm_uframes_t i_period_size;
- int i_channels;
+ /* Print some potentially useful debug */
+ msg_Dbg (aout, "using ALSA device: %s", sys->device);
+ DumpDevice (VLC_OBJECT(aout), pcm);
- if (spdif)
- {
- fourcc = VLC_CODEC_SPDIFL;
- i_buffer_size = ALSA_SPDIF_BUFFER_SIZE;
- pcm_format = SND_PCM_FORMAT_S16;
- i_channels = 2;
+ /* Get Initial hardware parameters */
+ snd_pcm_hw_params_t *hw;
+ unsigned param;
- p_aout->i_nb_samples = i_period_size = ALSA_SPDIF_PERIOD_SIZE;
- p_aout->format.i_bytes_per_frame = AOUT_SPDIF_SIZE;
- p_aout->format.i_frame_length = A52_FRAME_NB;
+ snd_pcm_hw_params_alloca (&hw);
+ snd_pcm_hw_params_any (pcm, hw);
+ Dump (aout, "initial hardware setup:\n", snd_pcm_hw_params_dump, hw);
- aout_VolumeNoneInit( p_aout );
- }
- else
+ val = snd_pcm_hw_params_set_rate_resample(pcm, hw, 0);
+ if (val)
{
- i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE;
- i_channels = aout_FormatNbChannels( &p_aout->format );
-
- p_aout->i_nb_samples = i_period_size = ALSA_DEFAULT_PERIOD_SIZE;
-
- aout_VolumeSoftInit( p_aout );
+ msg_Err (aout, "cannot disable resampling: %s", snd_strerror (val));
+ goto error;
}
- p_aout->pf_play = Play;
- p_aout->pf_pause = NULL;
-
- 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 )
+ val = snd_pcm_hw_params_set_access (pcm, hw,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (val)
{
- msg_Err( p_aout, "unable to retrieve hardware parameters (%s)",
- snd_strerror( val ) );
+ msg_Err (aout, "cannot set access mode: %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 )
+ /* Set sample format */
+ if (snd_pcm_hw_params_test_format (pcm, hw, pcm_format) == 0)
+ ;
+ else
+ if (snd_pcm_hw_params_test_format (pcm, hw, SND_PCM_FORMAT_FLOAT) == 0)
{
- msg_Err (p_aout, "cannot set sample format: %s", snd_strerror (val));
- goto error;
+ fmt->i_format = VLC_CODEC_FL32;
+ pcm_format = SND_PCM_FORMAT_FLOAT;
}
-
- val = snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw,
- SND_PCM_ACCESS_RW_INTERLEAVED );
- if( val < 0 )
+ else
+ if (snd_pcm_hw_params_test_format (pcm, hw, SND_PCM_FORMAT_S32) == 0)
{
- msg_Err( p_aout, "unable to set interleaved stream format (%s)",
- snd_strerror( val ) );
+ fmt->i_format = VLC_CODEC_S32N;
+ pcm_format = SND_PCM_FORMAT_S32;
+ }
+ else
+ if (snd_pcm_hw_params_test_format (pcm, hw, SND_PCM_FORMAT_S16) == 0)
+ {
+ fmt->i_format = VLC_CODEC_S16N;
+ pcm_format = SND_PCM_FORMAT_S16;
+ }
+ else
+ {
+ msg_Err (aout, "no supported sample format");
goto error;
}
- /* Set channels. */
- val = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm, p_hw, i_channels );
- if( val < 0 )
+ val = snd_pcm_hw_params_set_format (pcm, hw, pcm_format);
+ if (val)
{
- msg_Err( p_aout, "unable to set number of output channels (%s)",
- snd_strerror( val ) );
+ msg_Err (aout, "cannot set sample format: %s", snd_strerror (val));
goto error;
}
- /* Set rate. */
- unsigned rate = p_aout->format.i_rate;
- val = snd_pcm_hw_params_set_rate_near (p_sys->p_snd_pcm, p_hw, &rate,
- NULL);
- if (val < 0)
+ /* Set channels count */
+ unsigned channels;
+ if (!spdif)
{
- msg_Err (p_aout, "unable to set sampling rate (%s)",
+ sys->chans_to_reorder = SetupChannels (VLC_OBJECT(aout), pcm,
+ &fmt->i_physical_channels, sys->chans_table);
+ channels = popcount (fmt->i_physical_channels);
+ }
+ else
+ channels = 2;
+ fmt->i_original_channels = fmt->i_physical_channels;
+
+ /* By default, ALSA plug will pad missing channels with zeroes, which is
+ * usually fine. However, it will also discard extraneous channels, which
+ * is not acceptable. Thus the user must configure the physically
+ * available channels, and VLC will downmix if needed. */
+ val = snd_pcm_hw_params_set_channels (pcm, hw, channels);
+ if (val)
+ {
+ msg_Err (aout, "cannot set %u channels: %s", channels,
snd_strerror (val));
goto error;
}
- if (p_aout->format.i_rate != rate)
- msg_Warn (p_aout, "resampling from %d Hz to %d Hz",
- p_aout->format.i_rate, 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 )
+ /* Set sample rate */
+ val = snd_pcm_hw_params_set_rate_near (pcm, hw, &fmt->i_rate, NULL);
+ if (val)
{
- msg_Err( p_aout, "unable to set period size (%s)",
- snd_strerror( val ) );
+ msg_Err (aout, "cannot set sample rate: %s", snd_strerror (val));
goto error;
}
- p_aout->i_nb_samples = i_period_size;
+ sys->rate = fmt->i_rate;
- /* Set buffer size. */
- val = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm, p_hw,
- &i_buffer_size );
- if( val )
+ /* Set buffer size */
+ param = AOUT_MAX_ADVANCE_TIME;
+ val = snd_pcm_hw_params_set_buffer_time_near (pcm, hw, ¶m, NULL);
+ if (val)
{
- msg_Err( p_aout, "unable to set buffer size (%s)",
- snd_strerror( val ) );
+ msg_Err (aout, "cannot set buffer duration: %s", snd_strerror (val));
goto error;
}
-
- /* Commit hardware parameters. */
- val = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw );
- if( val < 0 )
+#if 0
+ val = snd_pcm_hw_params_get_buffer_time (hw, ¶m, NULL);
+ if (val)
{
- msg_Err( p_aout, "unable to commit hardware configuration (%s)",
- snd_strerror( val ) );
+ msg_Warn (aout, "cannot get buffer time: %s", snd_strerror(val));
+ param = AOUT_MIN_PREPARE_TIME;
+ }
+ else
+ param /= 2;
+#else /* work-around for period-long latency outputs (e.g. PulseAudio): */
+ param = AOUT_MIN_PREPARE_TIME;
+#endif
+ val = snd_pcm_hw_params_set_period_time_near (pcm, hw, ¶m, NULL);
+ if (val)
+ {
+ msg_Err (aout, "cannot set period: %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 )
+ /* Commit hardware parameters */
+ val = snd_pcm_hw_params (pcm, hw);
+ if (val < 0)
{
- msg_Err( p_aout, "unable to get period time (%s)",
- snd_strerror( val ) );
+ msg_Err (aout, "cannot commit hardware parameters: %s",
+ snd_strerror (val));
goto error;
}
+ Dump (aout, "final HW setup:\n", snd_pcm_hw_params_dump, hw);
/* Get Initial software parameters */
- snd_pcm_sw_params_current( p_sys->p_snd_pcm, p_sw );
+ snd_pcm_sw_params_t *sw;
+
+ snd_pcm_sw_params_alloca (&sw);
+ snd_pcm_sw_params_current (pcm, sw);
+ Dump (aout, "initial software parameters:\n", snd_pcm_sw_params_dump, sw);
- snd_pcm_sw_params_set_avail_min( p_sys->p_snd_pcm, p_sw,
- p_aout->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);
+ /* START REVISIT */
+ //snd_pcm_sw_params_set_avail_min( pcm, sw, i_period_size );
+ // FIXME: useful?
+ val = snd_pcm_sw_params_set_start_threshold (pcm, sw, 1);
if( val < 0 )
{
- msg_Err( p_aout, "unable to set start threshold (%s)",
+ msg_Err( aout, "unable to set start threshold (%s)",
snd_strerror( val ) );
goto error;
}
+ /* END REVISIT */
/* Commit software parameters. */
- if ( snd_pcm_sw_params( p_sys->p_snd_pcm, p_sw ) < 0 )
+ val = snd_pcm_sw_params (pcm, sw);
+ if (val)
{
- msg_Err( p_aout, "unable to set software configuration" );
+ msg_Err (aout, "cannot commit software parameters: %s",
+ snd_strerror (val));
goto error;
}
+ Dump (aout, "final software parameters:\n", snd_pcm_sw_params_dump, sw);
-#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 ) )
+ val = snd_pcm_prepare (pcm);
+ if (val)
{
- msg_Err( p_aout, "cannot create ALSA thread (%m)" );
- vlc_sem_destroy( &p_sys->wait );
+ msg_Err (aout, "cannot prepare device: %s", snd_strerror (val));
goto error;
}
- p_aout->format.i_format = fourcc;
- p_aout->format.i_rate = rate;
+ /* Setup audio_output_t */
+ if (spdif)
+ {
+ fmt->i_bytes_per_frame = AOUT_SPDIF_SIZE;
+ fmt->i_frame_length = A52_FRAME_NB;
+ }
+ else
+ {
+ aout_FormatPrepare (fmt);
+ sys->bits = fmt->i_bitspersample;
+ }
- Probe (obj);
+ aout->time_get = TimeGet;
+ aout->play = Play;
+ if (snd_pcm_hw_params_can_pause (hw))
+ aout->pause = Pause;
+ else
+ {
+ aout->pause = PauseDummy;
+ msg_Warn (aout, "device cannot be paused");
+ }
+ aout->flush = Flush;
+ aout_SoftVolumeStart (aout);
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 );
+ snd_pcm_close (pcm);
return VLC_EGENERIC;
}
-static void PlayIgnore( audio_output_t *p_aout )
-{ /* Already playing - nothing to do */
- (void) p_aout;
-}
-
-/*****************************************************************************
- * Play: start playback
- *****************************************************************************/
-static void Play( audio_output_t *p_aout )
+static int TimeGet (audio_output_t *aout, mtime_t *restrict delay)
{
- p_aout->pf_play = PlayIgnore;
+ aout_sys_t *sys = aout->sys;
+ snd_pcm_sframes_t frames;
- /* get the playing date of the first aout buffer */
- p_aout->sys->start_date = aout_FifoFirstDate( &p_aout->fifo );
-
- /* wake up the audio output thread */
- sem_post( &p_aout->sys->wait );
-}
-
-/*****************************************************************************
- * Close: close the ALSA device
- *****************************************************************************/
-static void Close (vlc_object_t *obj)
-{
- audio_output_t *p_aout = (audio_output_t *)obj;
- struct aout_sys_t * p_sys = p_aout->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 );
+ int val = snd_pcm_delay (sys->pcm, &frames);
+ if (val)
+ {
+ msg_Err (aout, "cannot estimate delay: %s", snd_strerror (val));
+ return -1;
+ }
+ *delay = frames * CLOCK_FREQ / sys->rate;
+ return 0;
}
-/*****************************************************************************
- * ALSAThread: asynchronous thread used to DMA the data to the device
- *****************************************************************************/
-static void* ALSAThread( void *data )
+/**
+ * Queues one audio buffer to the hardware.
+ */
+static void Play (audio_output_t *aout, block_t *block)
{
- audio_output_t * p_aout = data;
- struct aout_sys_t * p_sys = p_aout->sys;
+ aout_sys_t *sys = aout->sys;
- /* Wait for the exact time to start playing (avoids resampling) */
- vlc_sem_wait( &p_sys->wait );
- mwait( p_sys->start_date - AOUT_MAX_PTS_ADVANCE / 4 );
-#warning Should wait for buffer availability instead!
+ if (sys->chans_to_reorder != 0)
+ aout_ChannelReorder(block->p_buffer, block->i_buffer,
+ sys->chans_to_reorder, sys->chans_table, sys->bits);
- for(;;)
- ALSAFill( p_aout );
+ snd_pcm_t *pcm = sys->pcm;
- assert(0);
-}
-
-/*****************************************************************************
- * ALSAFill: function used to fill the ALSA buffer as much as possible
- *****************************************************************************/
-static void ALSAFill( audio_output_t * p_aout )
-{
- struct aout_sys_t * p_sys = p_aout->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;
+ /* TODO: better overflow handling */
+ /* TODO: no period wake ups */
- 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 )
+ while (block->i_nb_samples > 0)
{
- msg_Err( p_aout, "cannot get device status" );
- goto error;
- }
+ snd_pcm_sframes_t frames;
- /* 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 )
+ frames = snd_pcm_writei (pcm, block->p_buffer, block->i_nb_samples);
+ if (frames >= 0)
{
- msg_Err( p_aout, "cannot recover from buffer underrun" );
- goto error;
+ size_t bytes = snd_pcm_frames_to_bytes (pcm, frames);
+ block->i_nb_samples -= frames;
+ block->p_buffer += bytes;
+ block->i_buffer -= bytes;
+ // pts, length
}
-
- 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 )
+ else
{
- 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->format.i_bytes_per_frame
- / p_aout->format.i_rate
- * p_aout->format.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->format.i_bytes_per_frame );
- msg_Dbg( p_aout, "Rate: %d", p_aout->format.i_rate );
- msg_Dbg( p_aout, "Frame length: %d", p_aout->format.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->format.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 )
+ int val = snd_pcm_recover (pcm, frames, 1);
+ if (val)
+ {
+ msg_Err (aout, "cannot recover playback stream: %s",
+ snd_strerror (val));
+ DumpDeviceStatus (aout, pcm);
break;
+ }
+ msg_Warn (aout, "cannot write samples: %s", snd_strerror (frames));
}
-
- /* 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 );
-
}
+ block_Release (block);
+}
- 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 ) );
+/**
+ * Pauses/resumes the audio playback.
+ */
+static void Pause (audio_output_t *aout, bool pause, mtime_t date)
+{
+ snd_pcm_t *pcm = aout->sys->pcm;
- vlc_restorecancel(canc);
- msleep(p_sys->i_period_time / 2);
+ int val = snd_pcm_pause (pcm, pause);
+ if (unlikely(val))
+ PauseDummy (aout, pause, date);
}
-/*****************************************************************************
- * 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 )
+static void PauseDummy (audio_output_t *aout, bool pause, mtime_t date)
{
- module_config_t *p_item;
- (void)newval;
- (void)oldval;
- (void)p_unused;
+ snd_pcm_t *pcm = aout->sys->pcm;
- p_item = config_FindConfig( p_this, psz_name );
- if( !p_item ) return VLC_SUCCESS;
+ /* Stupid device cannot pause. Discard samples. */
+ if (pause)
+ snd_pcm_drop (pcm);
+ else
+ snd_pcm_prepare (pcm);
+ (void) date;
+}
- /* Clear-up the current list */
- if( p_item->i_list )
- {
- int i;
+/**
+ * Flushes/drains the audio playback buffer.
+ */
+static void Flush (audio_output_t *aout, bool wait)
+{
+ snd_pcm_t *pcm = aout->sys->pcm;
- /* 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;
+ if (wait)
+ snd_pcm_drain (pcm);
+ else
+ snd_pcm_drop (pcm);
+ snd_pcm_prepare (pcm);
+}
- GetDevices( p_this, p_item );
- /* Signal change to the interface */
- p_item->b_dirty = true;
+/**
+ * Releases the audio output.
+ */
+static void Stop (audio_output_t *aout)
+{
+ aout_sys_t *sys = aout->sys;
+ snd_pcm_t *pcm = sys->pcm;
- return VLC_SUCCESS;
+ snd_pcm_drop (pcm);
+ snd_pcm_close (pcm);
}
-
-static void GetDevices (vlc_object_t *obj, module_config_t *item)
+/**
+ * Enumerates ALSA output devices.
+ */
+static int EnumDevices(vlc_object_t *obj, char const *varname,
+ char ***restrict idp, char ***restrict namep)
{
void **hints;
- msg_Dbg(obj, "Available ALSA PCM devices:");
-
+ msg_Dbg (obj, "Available ALSA PCM devices:");
if (snd_device_name_hint(-1, "pcm", &hints) < 0)
- return;
+ return -1;
+
+ char **ids = NULL, **names = NULL;
+ unsigned n = 0;
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);
+ 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);
- }
+ ids = xrealloc (ids, (n + 1) * sizeof (*ids));
+ names = xrealloc (names, (n + 1) * sizeof (*names));
+ ids[n] = name;
+ names[n] = desc;
+ n++;
}
snd_device_name_free_hint(hints);
+ *idp = ids;
+ *namep = names;
+ (void) varname;
+ return n;
+}
- if (item != NULL)
- {
- item->ppsz_list[item->i_list] = NULL;
- item->ppsz_list_text[item->i_list] = NULL;
- }
+static int DevicesEnum (audio_output_t *aout, char ***idp, char ***namep)
+{
+ return EnumDevices (VLC_OBJECT(aout), NULL, idp, namep);
+}
+
+static int DeviceSelect (audio_output_t *aout, const char *id)
+{
+ aout_sys_t *sys = aout->sys;
+
+ char *device = strdup (id ? id : "default");
+ if (unlikely(device == NULL))
+ return -1;
+
+ free (sys->device);
+ sys->device = device;
+ aout_DeviceReport (aout, device);
+ aout_RestartRequest (aout, AOUT_RESTART_OUTPUT);
+ return 0;
+}
+
+static int Open(vlc_object_t *obj)
+{
+ audio_output_t *aout = (audio_output_t *)obj;
+ aout_sys_t *sys = malloc (sizeof (*sys));
+
+ if (unlikely(sys == NULL))
+ return VLC_ENOMEM;
+ sys->device = var_InheritString (aout, "alsa-audio-device");
+ if (unlikely(sys->device == NULL))
+ goto error;
+
+ aout->sys = sys;
+ aout->start = Start;
+ aout->stop = Stop;
+ aout_SoftVolumeInit (aout);
+ aout->device_enum = DevicesEnum;
+ aout->device_select = DeviceSelect;
+ aout_DeviceReport (aout, sys->device);
+ return VLC_SUCCESS;
+error:
+ free (sys);
+ return VLC_ENOMEM;
+}
+
+static void Close(vlc_object_t *obj)
+{
+ audio_output_t *aout = (audio_output_t *)obj;
+ aout_sys_t *sys = aout->sys;
+
+ free (sys->device);
+ free (sys);
}