/*****************************************************************************
* alsa.c : alsa plugin for vlc
*****************************************************************************
- * Copyright (C) 2000-2001 the VideoLAN team
- * $Id$
+ * Copyright (C) 2000-2010 the VideoLAN team
+ * 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
* 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;
};
#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 Open (vlc_object_t *);
+static void Close (vlc_object_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 * );
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_string ("alsa-audio-device", "default", N_("ALSA device"), 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_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;
+
+/** Helper for ALSA -> VLC debugging output */
+static void Dump (vlc_object_t *obj, const char *msg,
+ int (*cb)(void *, snd_output_t *), void *p)
+{
+ snd_output_t *output;
+ char *str;
+
+ if (unlikely(snd_output_buffer_open (&output)))
+ return;
+
+ 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;
+
+ 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));
+ }
+}
+
+static void DumpDeviceStatus (vlc_object_t *obj, snd_pcm_t *pcm)
+{
+ snd_pcm_status_t *status;
+
+ 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)
/**
* Initializes list of devices.
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 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 Open (vlc_object_t *obj)
{
- aout_instance_t * p_aout = (aout_instance_t *)obj;
+ audio_output_t *aout = (audio_output_t *)obj;
/* Get device name */
- char *psz_device;
+ char *device;
- if (var_Type (p_aout, "audio-device"))
- psz_device = var_GetString (p_aout, "audio-device");
+ if (var_Type (aout, "audio-device"))
+ device = var_GetString (aout, "audio-device");
else
- psz_device = var_InheritString( p_aout, "alsa-audio-device" );
- if (unlikely(psz_device == NULL))
+ device = var_InheritString (aout, "alsa-audio-device");
+ if (unlikely(device == NULL))
return VLC_ENOMEM;
snd_pcm_format_t pcm_format; /* ALSA sample format */
- vlc_fourcc_t fourcc = p_aout->output.output.i_format;
+ vlc_fourcc_t fourcc = aout->format.i_format;
bool spdif = false;
switch (fourcc)
pcm_format = SND_PCM_FORMAT_U8;
break;
default:
- if (AOUT_FMT_NON_LINEAR(&p_aout->output.output))
- spdif = var_InheritBool (p_aout, "spdif");
+ if (AOUT_FMT_SPDIF(&aout->format))
+ spdif = var_InheritBool (aout, "spdif");
if (HAVE_FPU)
{
fourcc = VLC_CODEC_FL32;
}
/* 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))
+ 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 (device, "default"))
{
unsigned aes3;
- switch (p_aout->output.output.i_rate)
+ switch (aout->format.i_rate)
{
#define FS(freq) \
case freq: aes3 = IEC958_AES3_CON_FS_ ## freq; break;
break;
}
- free (psz_device);
- if (asprintf (&psz_device,
+ free (device);
+ if (asprintf (&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,
}
/* Allocate structures */
- aout_sys_t *p_sys = malloc (sizeof (*p_sys));
- if (unlikely(p_sys == NULL))
+ aout_sys_t *sys = malloc (sizeof (*sys));
+ if (unlikely(sys == NULL))
{
- free (psz_device);
+ free (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
+ aout->sys = sys;
/* 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);
+ snd_pcm_t *pcm;
+ /* VLC always has a resampler. No need for ALSA's. */
+ const int mode = SND_PCM_NO_AUTO_RESAMPLE
+ /* ALSA discards extra channels (by default). This is not good. */
+ | SND_PCM_NO_AUTO_CHANNELS
+ /* VLC is currently unable to leverage ALSA softvol. No need for it. */
+ | SND_PCM_NO_SOFTVOL;
+
+ int val = snd_pcm_open (&pcm, 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",
+ var_Create (aout->p_libvlc, "alsa-working", VLC_VAR_BOOL);
+ if (val != 0 && var_GetBool (aout->p_libvlc, "alsa-working"))
+ dialog_Fatal (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);
+ var_SetBool (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_Create (aout->p_libvlc, "alsa-broken", VLC_VAR_BOOL);
+ if (!var_GetBool (aout->p_libvlc, "alsa-broken"))
{
- var_SetBool (p_aout->p_libvlc, "alsa-broken", true);
- dialog_Fatal (p_aout, "Potential ALSA version problem",
+ var_SetBool (aout->p_libvlc, "alsa-broken", true);
+ dialog_Fatal (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", 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);
+ device, snd_strerror (val));
+ free (device);
+ free (sys);
return VLC_EGENERIC;
}
- free( psz_device );
+ sys->pcm = pcm;
+
+ /* Print some potentially useful debug */
+ msg_Dbg (aout, "using ALSA device: %s", device);
+ free (device);
+ DumpDevice (VLC_OBJECT(aout), pcm);
- snd_pcm_uframes_t i_buffer_size;
- snd_pcm_uframes_t i_period_size;
- int i_channels;
+ /* Setup */
+ unsigned channels = aout_FormatNbChannels (&aout->format);
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 );
+ channels = 2;
}
- 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;
+ /* Get Initial hardware parameters */
+ snd_pcm_hw_params_t *hw;
+ unsigned param;
- snd_pcm_hw_params_alloca(&p_hw);
- snd_pcm_sw_params_alloca(&p_sw);
+ 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);
- /* Get Initial hardware parameters */
- val = snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw );
- if( val < 0 )
+ /* Set sample format */
+ val = snd_pcm_hw_params_set_format (pcm, hw, pcm_format);
+ if (val < 0)
{
- msg_Err( p_aout, "unable to retrieve hardware parameters (%s)",
- snd_strerror( val ) );
+ /* TODO: fallback to FL32 / S16N */
+ msg_Err (aout, "cannot set sample format: %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 )
+ val = snd_pcm_hw_params_set_access (pcm, hw,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (val)
{
- msg_Err (p_aout, "cannot set sample format: %s", snd_strerror (val));
+ msg_Err (aout, "cannot set access mode: %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 )
+ /* Set channels count */
+ val = snd_pcm_hw_params_set_channels (pcm, hw, channels);
+ if (val && channels > 2) /* Fallback to stereo */
{
- msg_Err( p_aout, "unable to set interleaved stream format (%s)",
- snd_strerror( val ) );
+ val = snd_pcm_hw_params_set_channels (pcm, hw, 2);
+ channels = 2;
+ }
+ if (val)
+ {
+ msg_Err (aout, "cannot set channels count: %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 )
+ /* Set sample rate */
+ unsigned rate = aout->format.i_rate;
+ val = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, NULL);
+ if (val)
{
- msg_Err( p_aout, "unable to set number of output channels (%s)",
- snd_strerror( val ) );
+ msg_Err (aout, "cannot set sample rate: %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 (aout->format.i_rate != rate)
+ msg_Dbg (aout, "resampling from %d Hz to %d Hz",
+ aout->format.i_rate, rate);
+
+ /* Set buffer size */
+ param = AOUT_MAX_ADVANCE_TIME;
+ val = snd_pcm_hw_params_set_buffer_time_near (pcm, hw, ¶m, NULL);
+ if (val)
+ msg_Warn (aout, "cannot set buffer duration near %u us: %s",
+ param, snd_strerror (val));
+ val = snd_pcm_hw_params_set_buffer_time_last (pcm, hw, ¶m, NULL);
+ if (val)
+ msg_Warn (aout, "cannot set buffer duration: %s", snd_strerror (val));
+
+ /* Set number of periods (at least two) */
+ param = 2;
+ val = snd_pcm_hw_params_set_periods_min (pcm, hw, ¶m, NULL);
+ if (val)
+ msg_Warn (aout, "cannot set minimum of %u periods: %s", param,
+ snd_strerror (val));
+ val = snd_pcm_hw_params_set_periods_first (pcm, hw, ¶m, NULL);
+ if (val)
+ msg_Warn (aout, "cannot set periods count near %u: %s", param,
+ snd_strerror (val));
+
+ /* Commit hardware parameters */
+ val = snd_pcm_hw_params (pcm, hw);
if (val < 0)
{
- msg_Err (p_aout, "unable to set sampling rate (%s)",
+ msg_Err (aout, "cannot commit hardware parameters: %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);
+ Dump (aout, "final HW setup:\n", snd_pcm_hw_params_dump, hw);
- /* 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;
+ /* Get Initial software parameters */
+ snd_pcm_sw_params_t *sw;
- /* 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;
- }
+ 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);
- /* Commit hardware parameters. */
- val = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw );
+ /* 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 commit hardware configuration (%s)",
+ msg_Err( aout, "unable to set start threshold (%s)",
snd_strerror( val ) );
goto error;
}
+ /* END REVISIT */
- val = snd_pcm_hw_params_get_period_time( p_hw, &p_sys->i_period_time,
- NULL );
- if( val < 0 )
+ /* Commit software parameters. */
+ val = snd_pcm_sw_params (pcm, sw);
+ if (val)
{
- msg_Err( p_aout, "unable to get period time (%s)",
- snd_strerror( val ) );
+ 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);
- /* 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 )
+ val = snd_pcm_prepare (pcm);
+ if (val)
{
- msg_Err( p_aout, "unable to set start threshold (%s)",
- snd_strerror( val ) );
+ msg_Err (aout, "cannot prepare device: %s", snd_strerror (val));
goto error;
}
- /* Commit software parameters. */
- if ( snd_pcm_sw_params( p_sys->p_snd_pcm, p_sw ) < 0 )
+ /* Setup audio_output_t */
+ aout->format.i_format = fourcc;
+ aout->format.i_rate = rate;
+ if (channels == 2)
+ aout->format.i_physical_channels = AOUT_CHAN_LEFT|AOUT_CHAN_RIGHT;
+ if (spdif)
{
- msg_Err( p_aout, "unable to set software configuration" );
- goto error;
+ aout->format.i_bytes_per_frame = AOUT_SPDIF_SIZE;
+ aout->format.i_frame_length = A52_FRAME_NB;
+ aout_VolumeNoneInit (aout);
}
+ else
+ aout_VolumeSoftInit (aout);
-#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 ) )
+ aout->pf_play = Play;
+ if (snd_pcm_hw_params_can_pause (hw))
+ aout->pf_pause = Pause;
+ else
{
- msg_Err( p_aout, "cannot create ALSA thread (%m)" );
- vlc_sem_destroy( &p_sys->wait );
- goto error;
+ aout->pf_pause = PauseDummy;
+ msg_Warn (aout, "device cannot be paused");
}
+ aout->pf_flush = Flush;
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 );
+ snd_pcm_close (pcm);
+ free (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 )
+/**
+ * Queues one audio buffer to the hardware.
+ */
+static void Play (audio_output_t *aout, block_t *block)
{
- 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;
- }
+ aout_sys_t *sys = aout->sys;
+ snd_pcm_t *pcm = sys->pcm;
+ snd_pcm_sframes_t frames;
+ snd_pcm_state_t state = snd_pcm_state (pcm);
- /* Handle buffer underruns and get the status again */
- if( snd_pcm_status_get_state( p_status ) == SND_PCM_STATE_XRUN )
+ if (snd_pcm_delay (pcm, &frames) == 0)
{
- /* 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;
- }
+ mtime_t delay = frames * CLOCK_FREQ / aout->format.i_rate;
- 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 )
+ if (state != SND_PCM_STATE_RUNNING)
{
- msg_Err( p_aout, "cannot get device status after recovery" );
- goto error;
+ delay = block->i_pts - (mdate () + delay);
+ if (delay > 0)
+ {
+ frames = (delay * aout->format.i_rate) / CLOCK_FREQ;
+ msg_Dbg (aout, "prepending %ld zeroes", frames);
+
+ void *pad = calloc (frames, aout->format.i_bytes_per_frame);
+ if (likely(pad != NULL))
+ {
+ snd_pcm_writei (pcm, pad, frames);
+ free (pad);
+ }
+ }
}
-
- /* 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;
+ else
+ aout_TimeReport (aout, block->i_pts - delay);
}
- 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;
+ /* TODO: better overflow handling */
+ /* TODO: no period wake ups */
- block_cleanup_push( p_buffer );
- for (;;)
+ while (block->i_nb_samples > 0)
{
- 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
+ frames = snd_pcm_writei (pcm, block->p_buffer, block->i_nb_samples);
+ if (frames >= 0)
{
- vlc_restorecancel(canc);
- poll(ufd, n, -1);
- canc = vlc_savecancel();
- snd_pcm_poll_descriptors_revents(p_pcm, ufd, n, &revents);
+ 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
}
- while(!revents);
-
- if(revents & POLLOUT)
+ else
{
- 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));
}
+ }
+ block_Release (block);
+}
- /* a suspend event occurred
- * (stream is suspended and waiting for an application recovery) */
- msg_Dbg( p_aout, "entering in suspend mode, trying to resume..." );
+/**
+ * Pauses/resumes the audio playback.
+ */
+static void Pause (audio_output_t *aout, bool pause, mtime_t date)
+{
+ snd_pcm_t *pcm = aout->sys->pcm;
- while( ( i_snd_rc = snd_pcm_resume( p_pcm ) ) == -EAGAIN )
- {
- vlc_restorecancel(canc);
- msleep(CLOCK_FREQ); /* device still suspended, wait... */
- canc = vlc_savecancel();
- }
+ int val = snd_pcm_pause (pcm, pause);
+ if (unlikely(val))
+ PauseDummy (aout, pause, date);
+}
- if( i_snd_rc < 0 )
- /* Device does not support resuming, restart it */
- i_snd_rc = snd_pcm_prepare( p_pcm );
+static void PauseDummy (audio_output_t *aout, bool pause, mtime_t date)
+{
+ snd_pcm_t *pcm = aout->sys->pcm;
- }
+ /* Stupid device cannot pause. Discard samples. */
+ if (pause)
+ snd_pcm_drop (pcm);
+ else
+ snd_pcm_prepare (pcm);
+ (void) date;
+}
+
+/**
+ * Flushes/drains the audio playback buffer.
+ */
+static void Flush (audio_output_t *aout, bool wait)
+{
+ snd_pcm_t *pcm = aout->sys->pcm;
- if( i_snd_rc < 0 )
- msg_Err( p_aout, "cannot write: %s", snd_strerror( i_snd_rc ) );
+ if (wait)
+ snd_pcm_drain (pcm);
+ else
+ snd_pcm_drop (pcm);
+ snd_pcm_prepare (pcm);
+}
- 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 ) );
+/**
+ * Releases the audio output.
+ */
+static void Close (vlc_object_t *obj)
+{
+ audio_output_t *aout = (audio_output_t *)obj;
+ aout_sys_t *sys = aout->sys;
+ snd_pcm_t *pcm = aout->sys->pcm;
- vlc_restorecancel(canc);
- msleep(p_sys->i_period_time / 2);
+ snd_pcm_drop (pcm);
+ snd_pcm_close (pcm);
+ free (sys);
}
/*****************************************************************************