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Commit 3d9f4dab authored by Arnaud de Bossoreille de Ribou's avatar Arnaud de Bossoreille de Ribou
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New ALSA audio output. It works with ALSA 0.9.0rc4 and earlier versions.

parent 1edaf53a
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Showing with 310 additions and 132 deletions
configure.ac.in
View file @ 3d9f4dab
......@@ -2023,7 +2023,7 @@ AC_ARG_ENABLE(alsa,
AC_CHECK_HEADER(alsa/asoundlib.h, AC_CHECK_LIB(asound, main, have_alsa="true", have_alsa="false"),have_alsa="false")
if test "x${have_alsa}" = "xtrue"
then
#PLUGINS="${PLUGINS} alsa"
PLUGINS="${PLUGINS} alsa"
LDFLAGS_alsa="${LDFLAGS_alsa} -lasound -lm -ldl"
fi
fi])
......
modules/audio_output/alsa.c
View file @ 3d9f4dab
......@@ -2,7 +2,7 @@
* alsa.c : alsa plugin for vlc
*****************************************************************************
* Copyright (C) 2000-2001 VideoLAN
* $Id: alsa.c,v 1.15 2002/10/22 20:55:27 sam Exp $
* $Id: alsa.c,v 1.16 2002/12/11 17:27:29 bozo Exp $
*
* Authors: Henri Fallon <henri@videolan.org> - Original Author
* Jeffrey Baker <jwbaker@acm.org> - Port to ALSA 1.0 API
......@@ -37,7 +37,10 @@
#include "aout_internal.h"
/* ALSA part */
/* ALSA part
Note: we use the new API which is available since 0.9.0rc4. */
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>
/*****************************************************************************
......@@ -50,7 +53,7 @@ struct aout_sys_t
{
snd_pcm_t * p_snd_pcm;
int i_period_time;
mtime_t grut;
#ifdef DEBUG
snd_output_t * p_snd_stderr;
#endif
......@@ -67,6 +70,9 @@ struct aout_sys_t
#define ALSA_SPDIF_PERIOD_SIZE A52_FRAME_NB
#define ALSA_SPDIF_BUFFER_SIZE ( ALSA_SPDIF_PERIOD_SIZE << 4 )
/* Why << 4 ? --Meuuh */
/* Why not ? --Bozo */
#define DEFAULT_ALSA_DEVICE "default"
/*****************************************************************************
* Local prototypes
......@@ -82,25 +88,143 @@ static void ALSAFill ( aout_instance_t * );
*****************************************************************************/
vlc_module_begin();
add_category_hint( N_("ALSA"), NULL );
add_string( "alsa-device", "default", aout_FindAndRestart,
N_("device name"), NULL );
add_string( "alsadev", DEFAULT_ALSA_DEVICE, aout_FindAndRestart,
N_("ALSA device name"), NULL );
set_description( _("ALSA audio module") );
set_capability( "audio output", 50 );
set_callbacks( Open, Close );
vlc_module_end();
/*****************************************************************************
* Probe: probe the audio device for available formats and channels
*****************************************************************************/
static void Probe( aout_instance_t * p_aout,
const char * psz_device, const char * psz_iec_device,
int i_snd_pcm_format )
{
struct aout_sys_t * p_sys = p_aout->output.p_sys;
vlc_value_t val;
var_Create ( p_aout, "audio-device", VLC_VAR_STRING | VLC_VAR_HASCHOICE );
/* Test for S/PDIF device if needed */
if ( psz_iec_device )
{
/* Opening the device should be enough */
if ( !snd_pcm_open( &p_sys->p_snd_pcm, psz_iec_device,
SND_PCM_STREAM_PLAYBACK, 0 ) )
{
val.psz_string = N_("S/PDIF");
var_Change( p_aout, "audio-device", VLC_VAR_ADDCHOICE, &val );
snd_pcm_close( p_sys->p_snd_pcm );
}
}
/* Now test linear PCM capabilities */
if ( !snd_pcm_open( &p_sys->p_snd_pcm, psz_device,
SND_PCM_STREAM_PLAYBACK, 0 ) )
{
int i_channels;
snd_pcm_hw_params_t * p_hw;
snd_pcm_hw_params_alloca (&p_hw);
if ( snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ) < 0 )
{
msg_Warn( p_aout, "unable to retrieve initial hardware parameters"
", disabling linear PCM audio" );
snd_pcm_close( p_sys->p_snd_pcm );
return;
}
if ( snd_pcm_hw_params_set_format( p_sys->p_snd_pcm, p_hw,
i_snd_pcm_format ) < 0 )
{
/* Assume a FPU enabled computer can handle float32 format.
If somebody tells us it's not always true then we'll have
to change this */
msg_Warn( p_aout, "unable to set stream sample size and word order"
", disabling linear PCM audio" );
snd_pcm_close( p_sys->p_snd_pcm );
return;
}
i_channels = aout_FormatNbChannels( &p_aout->output.output );
while ( i_channels > 1 )
{
/* Here we have to probe multi-channel capabilities but I have
no idea (at the moment) of how its managed by the ALSA
library.
It seems that '6' channels aren't well handled on a stereo
sound card like my i810 but it requires some more
investigations. That's why '4' and '6' cases are disabled.
-- Bozo */
if ( !snd_pcm_hw_params_test_channels( p_sys->p_snd_pcm, p_hw,
i_channels ) )
{
switch ( i_channels )
{
case 1:
val.psz_string = N_("Mono");
var_Change( p_aout, "audio-device",
VLC_VAR_ADDCHOICE, &val );
break;
case 2:
val.psz_string = N_("Stereo");
var_Change( p_aout, "audio-device",
VLC_VAR_ADDCHOICE, &val );
break;
/*
case 4:
val.psz_string = N_("2 Front 2 Rear");
var_Change( p_aout, "audio-device",
VLC_VAR_ADDCHOICE, &val );
break;
case 6:
val.psz_string = N_("5.1");
var_Change( p_aout, "audio-device",
VLC_VAR_ADDCHOICE, &val );
break;
*/
}
}
--i_channels;
}
/* Close the previously opened device */
snd_pcm_close( p_sys->p_snd_pcm );
}
/* Add final settings to the variable */
var_AddCallback( p_aout, "audio-device", aout_ChannelsRestart, NULL );
}
/*****************************************************************************
* Open: create a handle and open an alsa device
*****************************************************************************
* This function opens an alsa device, through the alsa API
* This function opens an alsa device, through the alsa API.
*
* Note: the only heap-allocated string is psz_device. All the other pointers
* are references to psz_device or to stack-allocated data.
*****************************************************************************/
static int Open( vlc_object_t *p_this )
{
aout_instance_t * p_aout = (aout_instance_t *)p_this;
struct aout_sys_t * p_sys;
char * psz_device;
int i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE;
int i_format = SND_PCM_FORMAT_S16;
vlc_value_t val;
char psz_default_iec_device[128]; /* Buffer used to store the default
S/PDIF device */
char * psz_device, * psz_iec_device; /* device names for PCM and S/PDIF
output */
int i_vlc_pcm_format; /* Audio format for VLC's data */
int i_snd_pcm_format; /* Audio format for ALSA's data */
snd_pcm_uframes_t i_buffer_size = 0;
snd_pcm_uframes_t i_period_size = 0;
int i_channels = 0;
snd_pcm_hw_params_t *p_hw;
snd_pcm_sw_params_t *p_sw;
......@@ -112,194 +236,243 @@ static int Open( vlc_object_t *p_this )
if( p_sys == NULL )
{
msg_Err( p_aout, "out of memory" );
return -1;
return VLC_ENOMEM;
}
/* Get device name */
if( (psz_device = config_GetPsz( p_aout, "dspdev" )) == NULL )
if( (psz_device = config_GetPsz( p_aout, "alsadev" )) == NULL )
{
msg_Err( p_aout, "no audio device given (maybe \"default\" ?)" );
free( p_sys );
return VLC_EGENERIC;
}
/* Choose the IEC device for S/PDIF output:
if the device is overriden by the user then it will be the one
otherwise we compute the default device based on the output format. */
if( AOUT_FMT_NON_LINEAR( &p_aout->output.output )
&& !strcmp( psz_device, DEFAULT_ALSA_DEVICE ) )
{
snprintf( psz_default_iec_device, sizeof(psz_default_iec_device),
"iec958:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x",
IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO,
IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
0,
( p_aout->output.output.i_rate == 48000 ?
IEC958_AES3_CON_FS_48000 :
( p_aout->output.output.i_rate == 44100 ?
IEC958_AES3_CON_FS_44100 : IEC958_AES3_CON_FS_32000 ) ) );
psz_iec_device = psz_default_iec_device;
}
else if( AOUT_FMT_NON_LINEAR( &p_aout->output.output ) )
{
psz_iec_device = psz_device;
}
else
{
psz_iec_device = NULL;
}
/* Choose the linear PCM format (read the comment above about FPU
and float32) */
if( p_aout->p_libvlc->i_cpu & CPU_CAPABILITY_FPU )
{
i_vlc_pcm_format = VLC_FOURCC('f','l','3','2');
i_snd_pcm_format = SND_PCM_FORMAT_FLOAT;
}
else
{
i_vlc_pcm_format = AOUT_FMT_S16_NE;
i_snd_pcm_format = SND_PCM_FORMAT_S16;
}
/* If the variable doesn't exist then it's the first time we're called
and we have to probe the available audio formats and channels */
if ( var_Type( p_aout, "audio-device" ) == 0 )
{
Probe( p_aout, psz_device, psz_iec_device, i_snd_pcm_format );
}
if ( var_Get( p_aout, "audio-device", &val ) < 0 )
{
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
if ( !strcmp( val.psz_string, N_("S/PDIF") ) )
{
p_aout->output.output.i_format = VLC_FOURCC('s','p','d','i');
}
else if ( !strcmp( val.psz_string, N_("5.1") ) )
{
p_aout->output.output.i_format = i_vlc_pcm_format;
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT
| AOUT_CHAN_LFE;
}
else if ( !strcmp( val.psz_string, N_("2 Front 2 Rear") ) )
{
p_aout->output.output.i_format = i_vlc_pcm_format;
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT;
}
else if ( !strcmp( val.psz_string, "Stereo" ) )
{
p_aout->output.output.i_format = i_vlc_pcm_format;
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
}
else if ( !strcmp( val.psz_string, "Mono" ) )
{
p_aout->output.output.i_format = i_vlc_pcm_format;
p_aout->output.output.i_physical_channels = AOUT_CHAN_CENTER;
}
free( val.psz_string );
#ifdef DEBUG
snd_output_stdio_attach( &p_sys->p_snd_stderr, stderr, 0 );
#endif
/* Open the device */
if ( AOUT_FMT_NON_LINEAR( &p_aout->output.output )
&& !strcmp( "default", psz_device ) )
if ( AOUT_FMT_NON_LINEAR( &p_aout->output.output ) )
{
/* ALSA doesn't understand "default" for S/PDIF. Cheat a little. */
char psz_iecdev[128];
if ( !strcmp( "default", psz_device ) )
if ( ( i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_iec_device,
SND_PCM_STREAM_PLAYBACK, 0 ) ) < 0 )
{
snprintf( psz_iecdev, sizeof(psz_iecdev),
"iec958:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x",
IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO,
IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
0,
(p_aout->output.output.i_rate == 48000 ?
IEC958_AES3_CON_FS_48000 :
(p_aout->output.output.i_rate == 44100 ?
IEC958_AES3_CON_FS_44100 : IEC958_AES3_CON_FS_32000)) );
}
else
{
strncat( psz_iecdev, psz_device, sizeof(psz_iecdev) );
}
if ( (i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_iecdev,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0 )
{
/* No S/PDIF. */
msg_Warn( p_aout, "cannot open S/PDIF ALSA device `%s' (%s)",
psz_device, snd_strerror(i_snd_rc) );
p_aout->output.output.i_format = VLC_FOURCC('f','l','3','2');
msg_Err( p_aout, "cannot open ALSA device `%s' (%s)",
psz_iec_device, snd_strerror( i_snd_rc ) );
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
else
{
i_buffer_size = ALSA_SPDIF_BUFFER_SIZE;
i_format = SND_PCM_FORMAT_S16;
i_buffer_size = ALSA_SPDIF_BUFFER_SIZE;
i_snd_pcm_format = SND_PCM_FORMAT_S16;
i_channels = 2;
p_aout->output.i_nb_samples = ALSA_SPDIF_PERIOD_SIZE;
p_aout->output.output.i_format = VLC_FOURCC('s','p','d','i');
p_aout->output.output.i_bytes_per_frame = AOUT_SPDIF_SIZE;
p_aout->output.output.i_frame_length = A52_FRAME_NB;
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 );
}
aout_VolumeNoneInit( p_aout );
}
if ( !AOUT_FMT_NON_LINEAR( &p_aout->output.output ) )
else
{
if ( (i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_device,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0 )
if ( ( i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_device,
SND_PCM_STREAM_PLAYBACK, 0 ) ) < 0 )
{
msg_Err( p_aout, "cannot open ALSA device `%s' (%s)",
psz_device, snd_strerror(i_snd_rc) );
psz_device, snd_strerror( i_snd_rc ) );
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
if ( p_aout->p_libvlc->i_cpu & CPU_CAPABILITY_FPU )
{
p_aout->output.output.i_format = VLC_FOURCC('f','l','3','2');
i_format = SND_PCM_FORMAT_FLOAT;
}
else
{
p_aout->output.output.i_format = AOUT_FMT_S16_NE;
i_format = SND_PCM_FORMAT_S16;
}
i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE;
p_aout->output.i_nb_samples = ALSA_DEFAULT_PERIOD_SIZE;
i_channels = aout_FormatNbChannels( &p_aout->output.output );
p_aout->output.i_nb_samples = i_period_size = ALSA_DEFAULT_PERIOD_SIZE;
aout_VolumeSoftInit( p_aout );
}
/* Free psz_device so that all the remaining data is stack-allocated */
free( psz_device );
p_aout->output.pf_play = Play;
snd_pcm_hw_params_alloca(&p_hw);
snd_pcm_sw_params_alloca(&p_sw);
if ( snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ) < 0 )
/* Get Initial hardware parameters */
if ( ( i_snd_rc = snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ) ) < 0 )
{
msg_Err( p_aout, "unable to retrieve initial hardware parameters" );
msg_Err( p_aout, "unable to retrieve initial hardware parameters (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set format. */
if ( snd_pcm_hw_params_set_format( p_sys->p_snd_pcm, p_hw, i_format ) < 0 )
if ( ( i_snd_rc = snd_pcm_hw_params_set_format( p_sys->p_snd_pcm, p_hw,
i_snd_pcm_format ) ) < 0 )
{
msg_Err( p_aout, "unable to set stream sample size and word order" );
msg_Err( p_aout, "unable to set stream sample size and word order (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
if ( !AOUT_FMT_NON_LINEAR( &p_aout->output.output ) )
if ( ( i_snd_rc = snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw,
SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
{
int i_nb_channels;
if ( snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw,
SND_PCM_ACCESS_RW_INTERLEAVED ) < 0 )
{
msg_Err( p_aout, "unable to set interleaved stream format" );
goto error;
}
/* Set channels. */
i_nb_channels = aout_FormatNbChannels( &p_aout->output.output );
msg_Err( p_aout, "unable to set interleaved stream format (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
if ( (i_snd_rc = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm,
p_hw, i_nb_channels )) < 0 )
{
msg_Err( p_aout, "unable to set number of output channels" );
goto error;
}
if ( i_snd_rc != i_nb_channels )
{
switch ( i_snd_rc )
{
case 1: p_aout->output.output.i_channels = AOUT_CHAN_MONO; break;
case 2: p_aout->output.output.i_channels = AOUT_CHAN_STEREO; break;
case 4: p_aout->output.output.i_channels = AOUT_CHAN_2F2R; break;
default:
msg_Err( p_aout, "Unsupported downmixing (%d)", i_snd_rc );
goto error;
}
}
/* Set channels. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm, p_hw,
i_channels ) ) < 0 )
{
msg_Err( p_aout, "unable to set number of output channels (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set rate. */
if ( (i_snd_rc = snd_pcm_hw_params_set_rate_near( p_sys->p_snd_pcm,
p_hw, p_aout->output.output.i_rate,
NULL )) < 0 )
{
msg_Err( p_aout, "unable to set sample rate" );
goto error;
}
p_aout->output.output.i_rate = i_snd_rc;
/* Set rate. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_rate_near( p_sys->p_snd_pcm, p_hw,
&p_aout->output.output.i_rate, NULL ) ) < 0 )
{
msg_Err( p_aout, "unable to set sample rate (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set buffer size. */
i_snd_rc = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm, p_hw,
i_buffer_size );
if( i_snd_rc < 0 )
if ( ( i_snd_rc = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm,
p_hw, &i_buffer_size ) ) < 0 )
{
msg_Err( p_aout, "unable to set buffer size" );
msg_Err( p_aout, "unable to set buffer size (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set period size. */
i_snd_rc = snd_pcm_hw_params_set_period_size_near(
p_sys->p_snd_pcm, p_hw, p_aout->output.i_nb_samples, NULL );
if( i_snd_rc < 0 )
if ( ( i_snd_rc = snd_pcm_hw_params_set_period_size_near( p_sys->p_snd_pcm,
p_hw, &i_period_size, NULL ) ) < 0 )
{
msg_Err( p_aout, "unable to set period size" );
msg_Err( p_aout, "unable to set period size (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
p_aout->output.i_nb_samples = i_snd_rc;
p_aout->output.i_nb_samples = i_period_size;
/* Write hardware configuration. */
if ( snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw ) < 0 )
/* Commit hardware parameters. */
if ( ( i_snd_rc = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw ) ) < 0 )
{
msg_Err( p_aout, "unable to set hardware configuration" );
msg_Err( p_aout, "unable to commit hardware configuration (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
p_sys->i_period_time = snd_pcm_hw_params_get_period_time( p_hw, NULL );
if( ( i_snd_rc = snd_pcm_hw_params_get_period_time( p_hw,
&p_sys->i_period_time, NULL ) ) < 0 )
{
msg_Err( p_aout, "unable to get period time (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Get Initial software parameters */
snd_pcm_sw_params_current( p_sys->p_snd_pcm, p_sw );
i_snd_rc = snd_pcm_sw_params_set_sleep_min( p_sys->p_snd_pcm, p_sw, 0 );
i_snd_rc = snd_pcm_sw_params_set_avail_min( p_sys->p_snd_pcm, p_sw,
p_aout->output.i_nb_samples );
/* Write software configuration. */
/* Commit software parameters. */
if ( snd_pcm_sw_params( p_sys->p_snd_pcm, p_sw ) < 0 )
{
msg_Err( p_aout, "unable to set software configuration" );
......@@ -382,6 +555,8 @@ static int ALSAThread( aout_instance_t * p_aout )
underruns */
/* Why do we need to sleep ? --Meuuh */
/* Maybe because I don't want to eat all the cpu by looping
all the time. --Bozo */
msleep( p_sys->i_period_time >> 2 );
}
......@@ -399,7 +574,6 @@ static void ALSAFill( aout_instance_t * p_aout )
snd_pcm_status_t * p_status;
snd_timestamp_t ts_next;
int i_snd_rc;
snd_pcm_uframes_t i_avail;
snd_pcm_status_alloca( &p_status );
......@@ -455,17 +629,16 @@ static void ALSAFill( aout_instance_t * p_aout )
/* Here the device should be either in the RUNNING state either in
the PREPARE state. p_status is valid. */
/* Try to write only if there is enough space */
i_avail = snd_pcm_status_get_avail( p_status );
if( i_avail >= p_aout->output.i_nb_samples )
if( 1 )
{
mtime_t next_date;
snd_pcm_status_get_tstamp( p_status, &ts_next );
next_date = (mtime_t)ts_next.tv_sec * 1000000 + ts_next.tv_usec;
next_date += (mtime_t)snd_pcm_status_get_delay(p_status)
* 1000000 / p_aout->output.output.i_rate;
p_buffer = aout_OutputNextBuffer( p_aout, next_date,
(p_aout->output.output.i_format !=
(p_aout->output.output.i_format ==
VLC_FOURCC('s','p','d','i')) );
/* Audio output buffer shortage -> stop the fill process and
......@@ -474,7 +647,7 @@ static void ALSAFill( aout_instance_t * p_aout )
return;
i_snd_rc = snd_pcm_writei( p_sys->p_snd_pcm, p_buffer->p_buffer,
p_buffer->i_nb_bytes );
p_buffer->i_nb_samples );
if( i_snd_rc < 0 )
{
......@@ -484,6 +657,11 @@ static void ALSAFill( aout_instance_t * p_aout )
aout_BufferFree( p_buffer );
}
else
{
/* Don't eat all the CPU. We will try to write later. */
return;
}
}
}
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