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Commit 5886e36b authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont
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ALSA: update to new audio output interface 

 * Buffer push straight from input, no more repacketization.
 * Support for pause/resume.
 * Support for flush (and unused support for drain).
 * Larger buffers if available (as with HDA cards).
 * Reduced periods count to minimum (they are not normally required).
parent c88c8eae
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Showing with 304 additions and 398 deletions
modules/audio_output/alsa.c
View file @ 5886e36b
......@@ -24,9 +24,6 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
......@@ -42,53 +39,19 @@
#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
{
aout_packet_t packet;
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 ( audio_output_t *, block_t * );
static void* ALSAThread ( void * );
static void ALSAFill ( audio_output_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 * );
......@@ -112,13 +75,13 @@ static const char *const ppsz_devices_text[] = {
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") )
......@@ -127,13 +90,57 @@ vlc_module_begin ()
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
/* ALSA just discards extra channels. Not good. Disable it. */
| SND_PCM_NO_AUTO_CHANNELS
/* 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.
......@@ -157,30 +164,29 @@ static void Probe (vlc_object_t *obj)
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)
{
audio_output_t * p_aout = (audio_output_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->format.i_format;
vlc_fourcc_t fourcc = aout->format.i_format;
bool spdif = false;
switch (fourcc)
......@@ -238,8 +244,8 @@ static int Open (vlc_object_t *obj)
pcm_format = SND_PCM_FORMAT_U8;
break;
default:
if (AOUT_FMT_SPDIF(&p_aout->format))
spdif = var_InheritBool (p_aout, "spdif");
if (AOUT_FMT_SPDIF(&aout->format))
spdif = var_InheritBool (aout, "spdif");
if (HAVE_FPU)
{
fourcc = VLC_CODEC_FL32;
......@@ -253,13 +259,13 @@ static int Open (vlc_object_t *obj)
}
/* 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->format.i_rate)
switch (aout->format.i_rate)
{
#define FS(freq) \
case freq: aes3 = IEC958_AES3_CON_FS_ ## freq; break;
......@@ -273,8 +279,8 @@ static int Open (vlc_object_t *obj)
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,
......@@ -283,439 +289,339 @@ static int Open (vlc_object_t *obj)
}
/* 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->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;
unsigned 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;
channels = 2;
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;
aout_VolumeNoneInit( p_aout );
}
else
{
i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE;
channels = aout_FormatNbChannels( &p_aout->format );
i_period_size = ALSA_DEFAULT_PERIOD_SIZE;
aout_VolumeSoftInit( p_aout );
}
p_aout->pf_play = Play;
p_aout->pf_pause = aout_PacketPause;
p_aout->pf_flush = aout_PacketFlush;
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;
}
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 ) );
goto error;
}
/* Set channels. */
val = snd_pcm_hw_params_set_channels (p_sys->p_snd_pcm, p_hw, channels);
if (val < 0 && channels > 2) /* Fallback to stereo */
{
val = snd_pcm_hw_params_set_channels (p_sys->p_snd_pcm, p_hw, 2);
val = snd_pcm_hw_params_set_channels (pcm, hw, 2);
channels = 2;
}
if (val < 0)
if (val)
{
msg_Err( p_aout, "unable to set number of output channels (%s)",
snd_strerror( val ) );
msg_Err (aout, "cannot set channels count: %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 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 sampling rate (%s)",
snd_strerror (val));
msg_Err (aout, "cannot set sample rate: %s", 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);
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, &param, 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, &param, 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, &param, 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, &param, NULL);
if (val)
msg_Warn (aout, "cannot set periods count near %u: %s", param,
snd_strerror (val));
/* 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 )
/* Commit hardware parameters */
val = snd_pcm_hw_params (pcm, hw);
if (val < 0)
{
msg_Err( p_aout, "unable to set period size (%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);
/* 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;
}
/* Get Initial software parameters */
snd_pcm_sw_params_t *sw;
/* Commit hardware parameters. */
val = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw );
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);
/* 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, i_period_size );
/* 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 );
aout_PacketInit (p_aout, &p_sys->packet, i_period_size);
/* 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)" );
aout_PacketDestroy (p_aout);
vlc_sem_destroy( &p_sys->wait );
goto error;
aout->pf_pause = PauseDummy;
msg_Warn (aout, "device cannot be paused");
}
p_aout->format.i_format = fourcc;
p_aout->format.i_rate = rate;
if (channels == 2)
p_aout->format.i_physical_channels = AOUT_CHAN_LEFT|AOUT_CHAN_RIGHT;
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;
}
/*****************************************************************************
* Play: start playback
*****************************************************************************/
static void Play( audio_output_t *p_aout, block_t *block )
{
/* get the playing date of the first aout buffer */
p_aout->sys->start_date = block->i_pts;
aout_PacketPlay( p_aout, block );
p_aout->pf_play = aout_PacketPlay;
/* 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 );
aout_PacketDestroy (p_aout);
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 )
{
audio_output_t * p_aout = data;
struct aout_sys_t * p_sys = p_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!
for(;;)
ALSAFill( p_aout );
assert(0);
}
/*****************************************************************************
* ALSAFill: function used to fill the ALSA buffer as much as possible
*****************************************************************************/
static void ALSAFill( audio_output_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->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;
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);
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 )
if (snd_pcm_delay (pcm, &frames) == 0)
{
msg_Err( p_aout, "cannot get device status" );
goto error;
}
mtime_t delay = frames * CLOCK_FREQ / aout->format.i_rate;
/* Handle buffer underruns and get the status again */
if( snd_pcm_status_get_state( p_status ) == SND_PCM_STATE_XRUN )
if (state != SND_PCM_STATE_RUNNING)
{
/* Prepare the device */
i_snd_rc = snd_pcm_prepare( p_pcm );
if( i_snd_rc )
delay = block->i_pts - (mdate () + delay);
if (delay > 0)
{
msg_Err( p_aout, "cannot recover from buffer underrun" );
goto error;
}
msg_Dbg( p_aout, "recovered from buffer underrun" );
frames = (delay * aout->format.i_rate) / CLOCK_FREQ;
msg_Dbg (aout, "prepending %ld zeroes", frames);
/* Get the new status */
i_snd_rc = snd_pcm_status( p_pcm, p_status );
if( i_snd_rc < 0 )
void *pad = calloc (frames, aout->format.i_bytes_per_frame);
if (likely(pad != NULL))
{
msg_Err( p_aout, "cannot get device status after recovery" );
goto error;
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->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;
aout_TimeReport (aout, block->i_pts - delay);
}
block_t *p_buffer = aout_PacketNext( p_aout, next_date );
/* TODO: better overflow handling */
/* TODO: no period wake ups */
/* Audio output buffer shortage -> stop the fill process and wait */
if( p_buffer == NULL )
goto error;
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
{
int val = snd_pcm_recover (pcm, frames, 1);
if (val)
{
i_snd_rc = snd_pcm_writei( p_pcm, p_buffer->p_buffer,
p_buffer->i_nb_samples );
if( i_snd_rc != -ESTRPIPE )
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;
}
if( i_snd_rc < 0 )
msg_Err( p_aout, "cannot write: %s", snd_strerror( i_snd_rc ) );
/**
* Flushes/drains the audio playback buffer.
*/
static void Flush (audio_output_t *aout, bool wait)
{
snd_pcm_t *pcm = aout->sys->pcm;
vlc_restorecancel(canc);
vlc_cleanup_run();
return;
if (wait)
snd_pcm_drain (pcm);
else
snd_pcm_drop (pcm);
snd_pcm_prepare (pcm);
}
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);
/**
* 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;
snd_pcm_drop (pcm);
snd_pcm_close (pcm);
free (sys);
}
/*****************************************************************************
......
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