1 /*****************************************************************************
2 * alsa.c : Alsa input module for vlc
3 *****************************************************************************
4 * Copyright (C) 2002-2009 the VideoLAN team
7 * Authors: Benjamin Pracht <bigben at videolan dot org>
8 * Richard Hosking <richard at hovis dot net>
9 * Antoine Cellerier <dionoea at videolan d.t org>
10 * Dennis Lou <dlou99 at yahoo dot com>
12 * This program is free software; you can redistribute it and/or modify
13 * it under the terms of the GNU General Public License as published by
14 * the Free Software Foundation; either version 2 of the License, or
15 * (at your option) any later version.
17 * This program is distributed in the hope that it will be useful,
18 * but WITHOUT ANY WARRANTY; without even the implied warranty of
19 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
20 * GNU General Public License for more details.
22 * You should have received a copy of the GNU General Public License
23 * along with this program; if not, write to the Free Software
24 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
25 *****************************************************************************/
28 * ALSA support based on parts of
29 * http://www.equalarea.com/paul/alsa-audio.html
30 * and hints taken from alsa-utils (aplay/arecord)
31 * http://www.alsa-project.org
34 /*****************************************************************************
36 *****************************************************************************/
42 #include <vlc_common.h>
43 #include <vlc_plugin.h>
44 #include <vlc_access.h>
45 #include <vlc_demux.h>
46 #include <vlc_input.h>
49 #include <sys/ioctl.h>
52 #include <sys/soundcard.h>
54 #define ALSA_PCM_NEW_HW_PARAMS_API
55 #define ALSA_PCM_NEW_SW_PARAMS_API
56 #include <alsa/asoundlib.h>
60 /*****************************************************************************
62 *****************************************************************************/
64 static int DemuxOpen ( vlc_object_t
* );
65 static void DemuxClose( vlc_object_t
* );
67 #define STEREO_TEXT N_( "Stereo" )
68 #define STEREO_LONGTEXT N_( \
69 "Capture the audio stream in stereo." )
71 #define SAMPLERATE_TEXT N_( "Samplerate" )
72 #define SAMPLERATE_LONGTEXT N_( \
73 "Samplerate of the captured audio stream, in Hz (eg: 11025, 22050, 44100, 48000)" )
75 #define CACHING_TEXT N_("Caching value in ms")
76 #define CACHING_LONGTEXT N_( \
77 "Caching value for Alsa captures. This " \
78 "value should be set in milliseconds." )
80 #define HELP_TEXT N_( \
81 "Use alsa:// to open the default audio input. If multiple audio " \
82 "inputs are available, they will be listed in the vlc debug output. " \
83 "To select hw:0,1 , use alsa://hw:0,1 ." )
85 #define ALSA_DEFAULT "hw"
86 #define CFG_PREFIX "alsa-"
89 set_shortname( N_("ALSA") )
90 set_description( N_("ALSA audio capture input") )
91 set_category( CAT_INPUT
)
92 set_subcategory( SUBCAT_INPUT_ACCESS
)
95 add_shortcut( "alsa" )
96 set_capability( "access_demux", 10 )
97 set_callbacks( DemuxOpen
, DemuxClose
)
99 add_bool( CFG_PREFIX
"stereo", true, NULL
, STEREO_TEXT
, STEREO_LONGTEXT
,
101 add_integer( CFG_PREFIX
"samplerate", 48000, NULL
, SAMPLERATE_TEXT
,
102 SAMPLERATE_LONGTEXT
, true )
103 add_integer( CFG_PREFIX
"caching", DEFAULT_PTS_DELAY
/ 1000, NULL
,
104 CACHING_TEXT
, CACHING_LONGTEXT
, true )
107 /*****************************************************************************
108 * Access: local prototypes
109 *****************************************************************************/
111 static int DemuxControl( demux_t
*, int, va_list );
113 static int Demux( demux_t
* );
115 static block_t
* GrabAudio( demux_t
*p_demux
);
117 static int OpenAudioDev( demux_t
*, const char * );
118 static bool ProbeAudioDevAlsa( demux_t
*, const char * );
119 static char *ListAvailableDevices( demux_t
*, bool b_probe
);
125 unsigned int i_sample_rate
;
127 size_t i_max_frame_size
;
132 snd_pcm_t
*p_alsa_pcm
;
133 size_t i_alsa_frame_size
;
134 int i_alsa_chunk_size
;
136 int64_t i_next_demux_date
; /* Used to handle alsa:// as input-slave properly */
139 static int FindMainDevice( demux_t
*p_demux
, const char *psz_device
)
143 msg_Dbg( p_demux
, "opening device '%s'", psz_device
);
144 if( ProbeAudioDevAlsa( p_demux
, psz_device
) )
146 msg_Dbg( p_demux
, "'%s' is an audio device", psz_device
);
147 OpenAudioDev( p_demux
, psz_device
);
150 else if( ProbeAudioDevAlsa( p_demux
, ALSA_DEFAULT
) )
152 msg_Dbg( p_demux
, "'%s' is an audio device", ALSA_DEFAULT
);
153 OpenAudioDev( p_demux
, ALSA_DEFAULT
);
155 else if( ( psz_device
= ListAvailableDevices( p_demux
, true ) ) )
157 msg_Dbg( p_demux
, "'%s' is an audio device", psz_device
);
158 OpenAudioDev( p_demux
, psz_device
);
159 free( (char *)psz_device
);
162 if( p_demux
->p_sys
->p_alsa_pcm
== NULL
)
167 static char *ListAvailableDevices( demux_t
*p_demux
, bool b_probe
)
169 snd_ctl_card_info_t
*p_info
= NULL
;
170 snd_ctl_card_info_alloca( &p_info
);
172 snd_pcm_info_t
*p_pcminfo
= NULL
;
173 snd_pcm_info_alloca( &p_pcminfo
);
176 msg_Dbg( p_demux
, "Available alsa capture devices:" );
178 while( !snd_card_next( &i_card
) && i_card
>= 0 )
180 char psz_devname
[10];
181 snprintf( psz_devname
, 10, "hw:%d", i_card
);
183 snd_ctl_t
*p_ctl
= NULL
;
184 if( snd_ctl_open( &p_ctl
, psz_devname
, 0 ) < 0 ) continue;
186 snd_ctl_card_info( p_ctl
, p_info
);
188 msg_Dbg( p_demux
, " %s (%s)",
189 snd_ctl_card_info_get_id( p_info
),
190 snd_ctl_card_info_get_name( p_info
) );
193 while( !snd_ctl_pcm_next_device( p_ctl
, &i_dev
) && i_dev
>= 0 )
195 snd_pcm_info_set_device( p_pcminfo
, i_dev
);
196 snd_pcm_info_set_subdevice( p_pcminfo
, 0 );
197 snd_pcm_info_set_stream( p_pcminfo
, SND_PCM_STREAM_CAPTURE
);
198 if( snd_ctl_pcm_info( p_ctl
, p_pcminfo
) < 0 ) continue;
201 msg_Dbg( p_demux
, " hw:%d,%d : %s (%s)", i_card
, i_dev
,
202 snd_pcm_info_get_id( p_pcminfo
),
203 snd_pcm_info_get_name( p_pcminfo
) );
207 if( asprintf( &psz_device
, "hw:%d,%d", i_card
, i_dev
) > 0 )
209 if( ProbeAudioDevAlsa( p_demux
, psz_device
) )
211 snd_ctl_close( p_ctl
);
220 snd_ctl_close( p_ctl
);
225 /*****************************************************************************
226 * DemuxOpen: opens alsa device, access_demux callback
227 *****************************************************************************
229 * url: <alsa device>::::
231 *****************************************************************************/
232 static int DemuxOpen( vlc_object_t
*p_this
)
234 demux_t
*p_demux
= (demux_t
*)p_this
;
237 /* Only when selected */
238 if( *p_demux
->psz_access
== '\0' ) return VLC_EGENERIC
;
241 p_demux
->pf_control
= DemuxControl
;
242 p_demux
->pf_demux
= Demux
;
243 p_demux
->info
.i_update
= 0;
244 p_demux
->info
.i_title
= 0;
245 p_demux
->info
.i_seekpoint
= 0;
247 p_demux
->p_sys
= p_sys
= calloc( 1, sizeof( demux_sys_t
) );
248 if( p_sys
== NULL
) return VLC_ENOMEM
;
250 p_sys
->i_sample_rate
= var_InheritInteger( p_demux
, CFG_PREFIX
"samplerate" );
251 p_sys
->b_stereo
= var_InheritBool( p_demux
, CFG_PREFIX
"stereo" );
252 p_sys
->i_cache
= var_InheritInteger( p_demux
, CFG_PREFIX
"caching" );
254 p_sys
->p_block
= NULL
;
255 p_sys
->i_next_demux_date
= -1;
257 const char *psz_device
= NULL
;
258 if( p_demux
->psz_location
&& *p_demux
->psz_location
)
259 psz_device
= p_demux
->psz_location
;
261 ListAvailableDevices( p_demux
, false );
263 if( FindMainDevice( p_demux
, psz_device
) != VLC_SUCCESS
)
265 if( p_demux
->psz_location
&& *p_demux
->psz_location
)
266 ListAvailableDevices( p_demux
, false );
267 DemuxClose( p_this
);
274 /*****************************************************************************
275 * Close: close device, free resources
276 *****************************************************************************/
277 static void DemuxClose( vlc_object_t
*p_this
)
279 demux_t
*p_demux
= (demux_t
*)p_this
;
280 demux_sys_t
*p_sys
= p_demux
->p_sys
;
282 if( p_sys
->p_alsa_pcm
)
284 snd_pcm_close( p_sys
->p_alsa_pcm
);
287 if( p_sys
->p_block
) block_Release( p_sys
->p_block
);
292 /*****************************************************************************
294 *****************************************************************************/
295 static int DemuxControl( demux_t
*p_demux
, int i_query
, va_list args
)
297 demux_sys_t
*p_sys
= p_demux
->p_sys
;
301 /* Special for access_demux */
302 case DEMUX_CAN_PAUSE
:
304 case DEMUX_SET_PAUSE_STATE
:
305 case DEMUX_CAN_CONTROL_PACE
:
306 *va_arg( args
, bool * ) = false;
309 case DEMUX_GET_PTS_DELAY
:
310 *va_arg( args
, int64_t * ) = (int64_t)p_sys
->i_cache
* 1000;
314 *va_arg( args
, int64_t * ) = mdate();
317 case DEMUX_SET_NEXT_DEMUX_TIME
:
318 p_sys
->i_next_demux_date
= va_arg( args
, int64_t );
321 /* TODO implement others */
329 /*****************************************************************************
330 * Demux: Processes the audio frame
331 *****************************************************************************/
332 static int Demux( demux_t
*p_demux
)
334 demux_sys_t
*p_sys
= p_demux
->p_sys
;
336 block_t
*p_block
= NULL
;
342 es_out_Send( p_demux
->out
, p_sys
->p_es
, p_block
);
347 int i_wait
= snd_pcm_wait( p_sys
->p_alsa_pcm
, 10 ); /* See poll() comment in oss.c */
352 p_block
= GrabAudio( p_demux
);
354 es_out_Control( p_demux
->out
, ES_OUT_SET_PCR
, p_block
->i_pts
);
357 /* FIXME: this is a copy paste from below. Shouldn't be needed
361 snd_pcm_prepare( p_sys
->p_alsa_pcm
);
366 int i_resume
= snd_pcm_resume( p_sys
->p_alsa_pcm
);
367 if( i_resume
< 0 && i_resume
!= -EAGAIN
) snd_pcm_prepare( p_sys
->p_alsa_pcm
);
372 } while( p_block
&& p_sys
->i_next_demux_date
> 0 &&
373 p_block
->i_pts
< p_sys
->i_next_demux_date
);
376 es_out_Send( p_demux
->out
, p_sys
->p_es
, p_block
);
382 /*****************************************************************************
383 * GrabAudio: Grab an audio frame
384 *****************************************************************************/
385 static block_t
* GrabAudio( demux_t
*p_demux
)
387 demux_sys_t
*p_sys
= p_demux
->p_sys
;
388 int i_read
, i_correct
;
391 if( p_sys
->p_block
) p_block
= p_sys
->p_block
;
392 else p_block
= block_New( p_demux
, p_sys
->i_max_frame_size
);
396 msg_Warn( p_demux
, "cannot get block" );
400 p_sys
->p_block
= p_block
;
403 i_read
= snd_pcm_readi( p_sys
->p_alsa_pcm
, p_block
->p_buffer
, p_sys
->i_alsa_chunk_size
);
413 snd_pcm_prepare( p_sys
->p_alsa_pcm
);
417 i_resume
= snd_pcm_resume( p_sys
->p_alsa_pcm
);
418 if( i_resume
< 0 && i_resume
!= -EAGAIN
) snd_pcm_prepare( p_sys
->p_alsa_pcm
);
421 msg_Err( p_demux
, "Failed to read alsa frame (%s)", snd_strerror( i_read
) );
427 /* convert from frames to bytes */
428 i_read
*= p_sys
->i_alsa_frame_size
;
431 if( i_read
<= 0 ) return 0;
433 p_block
->i_buffer
= i_read
;
436 /* Correct the date because of kernel buffering */
440 snd_pcm_sframes_t delay
= 0;
441 if( ( i_err
= snd_pcm_delay( p_sys
->p_alsa_pcm
, &delay
) ) >= 0 )
443 size_t i_correction_delta
= delay
* p_sys
->i_alsa_frame_size
;
444 /* Test for overrun */
445 if( i_correction_delta
> p_sys
->i_max_frame_size
)
447 msg_Warn( p_demux
, "ALSA read overrun (%zu > %zu)",
448 i_correction_delta
, p_sys
->i_max_frame_size
);
449 i_correction_delta
= p_sys
->i_max_frame_size
;
450 snd_pcm_prepare( p_sys
->p_alsa_pcm
);
452 i_correct
+= i_correction_delta
;
456 /* delay failed so reset */
457 msg_Warn( p_demux
, "ALSA snd_pcm_delay failed (%s)", snd_strerror( i_err
) );
458 snd_pcm_prepare( p_sys
->p_alsa_pcm
);
462 p_block
->i_pts
= p_block
->i_dts
=
463 mdate() - INT64_C(1000000) * (mtime_t
)i_correct
/
464 2 / ( p_sys
->b_stereo
? 2 : 1) / p_sys
->i_sample_rate
;
469 /*****************************************************************************
470 * OpenAudioDev: open and set up the audio device and probe for capabilities
471 *****************************************************************************/
472 static int OpenAudioDevAlsa( demux_t
*p_demux
, const char *psz_device
)
474 demux_sys_t
*p_sys
= p_demux
->p_sys
;
475 p_sys
->p_alsa_pcm
= NULL
;
476 snd_pcm_hw_params_t
*p_hw_params
= NULL
;
477 snd_pcm_uframes_t buffer_size
;
478 snd_pcm_uframes_t chunk_size
;
483 if( ( i_err
= snd_pcm_open( &p_sys
->p_alsa_pcm
, psz_device
,
484 SND_PCM_STREAM_CAPTURE
, SND_PCM_NONBLOCK
) ) < 0)
486 msg_Err( p_demux
, "Cannot open ALSA audio device %s (%s)",
487 psz_device
, snd_strerror( i_err
) );
491 if( ( i_err
= snd_pcm_nonblock( p_sys
->p_alsa_pcm
, 1 ) ) < 0)
493 msg_Err( p_demux
, "Cannot set ALSA nonblock (%s)",
494 snd_strerror( i_err
) );
498 /* Begin setting hardware parameters */
500 if( ( i_err
= snd_pcm_hw_params_malloc( &p_hw_params
) ) < 0 )
503 "ALSA: cannot allocate hardware parameter structure (%s)",
504 snd_strerror( i_err
) );
508 if( ( i_err
= snd_pcm_hw_params_any( p_sys
->p_alsa_pcm
, p_hw_params
) ) < 0 )
511 "ALSA: cannot initialize hardware parameter structure (%s)",
512 snd_strerror( i_err
) );
516 /* Set Interleaved access */
517 if( ( i_err
= snd_pcm_hw_params_set_access( p_sys
->p_alsa_pcm
, p_hw_params
, SND_PCM_ACCESS_RW_INTERLEAVED
) ) < 0 )
519 msg_Err( p_demux
, "ALSA: cannot set access type (%s)",
520 snd_strerror( i_err
) );
524 /* Set 16 bit little endian */
525 if( ( i_err
= snd_pcm_hw_params_set_format( p_sys
->p_alsa_pcm
, p_hw_params
, SND_PCM_FORMAT_S16_LE
) ) < 0 )
527 msg_Err( p_demux
, "ALSA: cannot set sample format (%s)",
528 snd_strerror( i_err
) );
532 /* Set sample rate */
533 i_err
= snd_pcm_hw_params_set_rate_near( p_sys
->p_alsa_pcm
, p_hw_params
, &p_sys
->i_sample_rate
, NULL
);
536 msg_Err( p_demux
, "ALSA: cannot set sample rate (%s)",
537 snd_strerror( i_err
) );
542 unsigned int channels
= p_sys
->b_stereo
? 2 : 1;
543 if( ( i_err
= snd_pcm_hw_params_set_channels( p_sys
->p_alsa_pcm
, p_hw_params
, channels
) ) < 0 )
545 channels
= ( channels
==1 ) ? 2 : 1;
546 msg_Warn( p_demux
, "ALSA: cannot set channel count (%s). "
547 "Trying with channels=%d",
548 snd_strerror( i_err
),
550 if( ( i_err
= snd_pcm_hw_params_set_channels( p_sys
->p_alsa_pcm
, p_hw_params
, channels
) ) < 0 )
552 msg_Err( p_demux
, "ALSA: cannot set channel count (%s)",
553 snd_strerror( i_err
) );
556 p_sys
->b_stereo
= ( channels
== 2 );
559 /* Set metrics for buffer calculations later */
560 unsigned int buffer_time
;
561 if( ( i_err
= snd_pcm_hw_params_get_buffer_time_max(p_hw_params
, &buffer_time
, 0) ) < 0 )
563 msg_Err( p_demux
, "ALSA: cannot get buffer time max (%s)",
564 snd_strerror( i_err
) );
567 if( buffer_time
> 500000 ) buffer_time
= 500000;
569 /* Set period time */
570 unsigned int period_time
= buffer_time
/ 4;
571 i_err
= snd_pcm_hw_params_set_period_time_near( p_sys
->p_alsa_pcm
, p_hw_params
, &period_time
, 0 );
574 msg_Err( p_demux
, "ALSA: cannot set period time (%s)",
575 snd_strerror( i_err
) );
579 /* Set buffer time */
580 i_err
= snd_pcm_hw_params_set_buffer_time_near( p_sys
->p_alsa_pcm
, p_hw_params
, &buffer_time
, 0 );
583 msg_Err( p_demux
, "ALSA: cannot set buffer time (%s)",
584 snd_strerror( i_err
) );
588 /* Apply new hardware parameters */
589 if( ( i_err
= snd_pcm_hw_params( p_sys
->p_alsa_pcm
, p_hw_params
) ) < 0 )
591 msg_Err( p_demux
, "ALSA: cannot set hw parameters (%s)",
592 snd_strerror( i_err
) );
596 /* Get various buffer metrics */
597 snd_pcm_hw_params_get_period_size( p_hw_params
, &chunk_size
, 0 );
598 snd_pcm_hw_params_get_buffer_size( p_hw_params
, &buffer_size
);
599 if( chunk_size
== buffer_size
)
602 "ALSA: period cannot equal buffer size (%lu == %lu)",
603 chunk_size
, buffer_size
);
607 int bits_per_sample
= snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE
);
608 int bits_per_frame
= bits_per_sample
* channels
;
610 p_sys
->i_alsa_chunk_size
= chunk_size
;
611 p_sys
->i_alsa_frame_size
= bits_per_frame
/ 8;
612 p_sys
->i_max_frame_size
= chunk_size
* bits_per_frame
/ 8;
614 snd_pcm_hw_params_free( p_hw_params
);
618 if( ( i_err
= snd_pcm_prepare( p_sys
->p_alsa_pcm
) ) < 0 )
621 "ALSA: cannot prepare audio interface for use (%s)",
622 snd_strerror( i_err
) );
626 snd_pcm_start( p_sys
->p_alsa_pcm
);
632 if( p_hw_params
) snd_pcm_hw_params_free( p_hw_params
);
633 if( p_sys
->p_alsa_pcm
) snd_pcm_close( p_sys
->p_alsa_pcm
);
634 p_sys
->p_alsa_pcm
= NULL
;
640 static int OpenAudioDev( demux_t
*p_demux
, const char *psz_device
)
642 demux_sys_t
*p_sys
= p_demux
->p_sys
;
643 if( OpenAudioDevAlsa( p_demux
, psz_device
) != VLC_SUCCESS
)
646 msg_Dbg( p_demux
, "opened adev=`%s' %s %dHz",
647 psz_device
, p_sys
->b_stereo
? "stereo" : "mono",
648 p_sys
->i_sample_rate
);
651 es_format_Init( &fmt
, AUDIO_ES
, VLC_FOURCC('a','r','a','w') );
653 fmt
.audio
.i_channels
= p_sys
->b_stereo
? 2 : 1;
654 fmt
.audio
.i_rate
= p_sys
->i_sample_rate
;
655 fmt
.audio
.i_bitspersample
= 16;
656 fmt
.audio
.i_blockalign
= fmt
.audio
.i_channels
* fmt
.audio
.i_bitspersample
/ 8;
657 fmt
.i_bitrate
= fmt
.audio
.i_channels
* fmt
.audio
.i_rate
* fmt
.audio
.i_bitspersample
;
659 msg_Dbg( p_demux
, "new audio es %d channels %dHz",
660 fmt
.audio
.i_channels
, fmt
.audio
.i_rate
);
662 p_sys
->p_es
= es_out_Add( p_demux
->out
, &fmt
);
667 /*****************************************************************************
668 * ProbeAudioDevAlsa: probe audio for capabilities
669 *****************************************************************************/
670 static bool ProbeAudioDevAlsa( demux_t
*p_demux
, const char *psz_device
)
673 snd_pcm_t
*p_alsa_pcm
;
675 if( ( i_err
= snd_pcm_open( &p_alsa_pcm
, psz_device
, SND_PCM_STREAM_CAPTURE
, SND_PCM_NONBLOCK
) ) < 0 )
677 msg_Err( p_demux
, "cannot open device %s for ALSA audio (%s)", psz_device
, snd_strerror( i_err
) );
681 snd_pcm_close( p_alsa_pcm
);