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>
52 #include <sys/ioctl.h>
55 #include <sys/soundcard.h>
57 #define ALSA_PCM_NEW_HW_PARAMS_API
58 #define ALSA_PCM_NEW_SW_PARAMS_API
59 #include <alsa/asoundlib.h>
63 /*****************************************************************************
65 *****************************************************************************/
67 static int DemuxOpen ( vlc_object_t * );
68 static void DemuxClose( vlc_object_t * );
70 #define STEREO_TEXT N_( "Stereo" )
71 #define STEREO_LONGTEXT N_( \
72 "Capture the audio stream in stereo." )
74 #define SAMPLERATE_TEXT N_( "Samplerate" )
75 #define SAMPLERATE_LONGTEXT N_( \
76 "Samplerate of the captured audio stream, in Hz (eg: 11025, 22050, 44100, 48000)" )
78 #define CACHING_TEXT N_("Caching value in ms")
79 #define CACHING_LONGTEXT N_( \
80 "Caching value for Alsa captures. This " \
81 "value should be set in milliseconds." )
83 #define ALSA_DEFAULT "hw"
84 #define CFG_PREFIX "alsa-"
87 set_shortname( N_("Alsa") )
88 set_description( N_("Alsa audio capture input") )
89 set_category( CAT_INPUT )
90 set_subcategory( SUBCAT_INPUT_ACCESS )
92 add_shortcut( "alsa" )
93 set_capability( "access_demux", 10 )
94 set_callbacks( DemuxOpen, DemuxClose )
96 add_bool( CFG_PREFIX "stereo", true, NULL, STEREO_TEXT, STEREO_LONGTEXT,
98 add_integer( CFG_PREFIX "samplerate", 48000, NULL, SAMPLERATE_TEXT,
99 SAMPLERATE_LONGTEXT, true )
100 add_integer( CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL,
101 CACHING_TEXT, CACHING_LONGTEXT, true )
104 /*****************************************************************************
105 * Access: local prototypes
106 *****************************************************************************/
108 static int DemuxControl( demux_t *, int, va_list );
110 static int Demux( demux_t * );
112 static block_t* GrabAudio( demux_t *p_demux );
114 static int OpenAudioDev( demux_t * );
115 static bool ProbeAudioDevAlsa( demux_t *, const char *psz_device );
119 const char *psz_device; /* Alsa device from MRL */
123 unsigned int i_sample_rate;
125 size_t i_max_frame_size;
130 snd_pcm_t *p_alsa_pcm;
131 size_t i_alsa_frame_size;
132 int i_alsa_chunk_size;
134 int64_t i_next_demux_date; /* Used to handle alsa:// as input-slave properly */
137 static int FindMainDevice( demux_t *p_demux )
139 /* TODO: if using default device, loop through all alsa devices until
141 msg_Dbg( p_demux, "opening device '%s'", p_demux->p_sys->psz_device );
142 if( ProbeAudioDevAlsa( p_demux, p_demux->p_sys->psz_device ) )
144 msg_Dbg( p_demux, "'%s' is an audio device",
145 p_demux->p_sys->psz_device );
146 OpenAudioDev( p_demux );
149 if( p_demux->p_sys->p_alsa_pcm == NULL )
154 /*****************************************************************************
155 * DemuxOpen: opens alsa device, access_demux callback
156 *****************************************************************************
158 * url: <alsa device>::::
160 *****************************************************************************/
161 static int DemuxOpen( vlc_object_t *p_this )
163 demux_t *p_demux = (demux_t*)p_this;
166 /* Only when selected */
167 if( *p_demux->psz_access == '\0' ) return VLC_EGENERIC;
170 p_demux->pf_control = DemuxControl;
171 p_demux->pf_demux = Demux;
172 p_demux->info.i_update = 0;
173 p_demux->info.i_title = 0;
174 p_demux->info.i_seekpoint = 0;
176 p_demux->p_sys = p_sys = calloc( 1, sizeof( demux_sys_t ) );
177 if( p_sys == NULL ) return VLC_ENOMEM;
179 p_sys->i_sample_rate = var_CreateGetInteger( p_demux, CFG_PREFIX "samplerate" );
180 p_sys->b_stereo = var_CreateGetBool( p_demux, CFG_PREFIX "stereo" );
181 p_sys->i_cache = var_CreateGetInteger( p_demux, CFG_PREFIX "caching" );
183 p_sys->p_block = NULL;
184 p_sys->i_next_demux_date = -1;
186 if( p_demux->psz_path && *p_demux->psz_path )
187 p_sys->psz_device = p_demux->psz_path;
189 p_sys->psz_device = ALSA_DEFAULT;
191 if( FindMainDevice( p_demux ) != VLC_SUCCESS )
193 DemuxClose( p_this );
200 /*****************************************************************************
201 * Close: close device, free resources
202 *****************************************************************************/
203 static void DemuxClose( vlc_object_t *p_this )
205 demux_t *p_demux = (demux_t *)p_this;
206 demux_sys_t *p_sys = p_demux->p_sys;
208 if( p_sys->p_alsa_pcm )
210 snd_pcm_close( p_sys->p_alsa_pcm );
213 if( p_sys->p_block ) block_Release( p_sys->p_block );
218 /*****************************************************************************
220 *****************************************************************************/
221 static int DemuxControl( demux_t *p_demux, int i_query, va_list args )
223 demux_sys_t *p_sys = p_demux->p_sys;
229 /* Special for access_demux */
230 case DEMUX_CAN_PAUSE:
232 case DEMUX_SET_PAUSE_STATE:
233 case DEMUX_CAN_CONTROL_PACE:
234 pb = (bool*)va_arg( args, bool * );
238 case DEMUX_GET_PTS_DELAY:
239 pi64 = (int64_t*)va_arg( args, int64_t * );
240 *pi64 = (int64_t)p_sys->i_cache * 1000;
244 pi64 = (int64_t*)va_arg( args, int64_t * );
248 case DEMUX_SET_NEXT_DEMUX_TIME:
249 p_sys->i_next_demux_date = (int64_t)va_arg( args, int64_t );
252 /* TODO implement others */
260 /*****************************************************************************
261 * Demux: Processes the audio frame
262 *****************************************************************************/
263 static int Demux( demux_t *p_demux )
265 demux_sys_t *p_sys = p_demux->p_sys;
267 block_t *p_block = NULL;
273 es_out_Send( p_demux->out, p_sys->p_es, p_block );
278 int i_wait = snd_pcm_wait( p_sys->p_alsa_pcm, 500 );
283 p_block = GrabAudio( p_demux );
285 es_out_Control( p_demux->out, ES_OUT_SET_PCR, p_block->i_pts );
288 /* FIXME: this is a copy paste from below. Shouldn't be needed
292 snd_pcm_prepare( p_sys->p_alsa_pcm );
297 int i_resume = snd_pcm_resume( p_sys->p_alsa_pcm );
298 if( i_resume < 0 && i_resume != -EAGAIN ) snd_pcm_prepare( p_sys->p_alsa_pcm );
303 } while( p_block && p_sys->i_next_demux_date > 0 &&
304 p_block->i_pts < p_sys->i_next_demux_date );
307 es_out_Send( p_demux->out, p_sys->p_es, p_block );
313 /*****************************************************************************
314 * GrabAudio: Grab an audio frame
315 *****************************************************************************/
316 static block_t* GrabAudio( demux_t *p_demux )
318 demux_sys_t *p_sys = p_demux->p_sys;
319 int i_read, i_correct;
322 if( p_sys->p_block ) p_block = p_sys->p_block;
323 else p_block = block_New( p_demux, p_sys->i_max_frame_size );
327 msg_Warn( p_demux, "cannot get block" );
331 p_sys->p_block = p_block;
334 i_read = snd_pcm_readi( p_sys->p_alsa_pcm, p_block->p_buffer, p_sys->i_alsa_chunk_size );
344 snd_pcm_prepare( p_sys->p_alsa_pcm );
348 i_resume = snd_pcm_resume( p_sys->p_alsa_pcm );
349 if( i_resume < 0 && i_resume != -EAGAIN ) snd_pcm_prepare( p_sys->p_alsa_pcm );
352 msg_Err( p_demux, "Failed to read alsa frame (%s)", snd_strerror( i_read ) );
358 /* convert from frames to bytes */
359 i_read *= p_sys->i_alsa_frame_size;
362 if( i_read <= 0 ) return 0;
364 p_block->i_buffer = i_read;
367 /* Correct the date because of kernel buffering */
371 snd_pcm_sframes_t delay = 0;
372 if( ( i_err = snd_pcm_delay( p_sys->p_alsa_pcm, &delay ) ) >= 0 )
374 size_t i_correction_delta = delay * p_sys->i_alsa_frame_size;
375 /* Test for overrun */
376 if( i_correction_delta > p_sys->i_max_frame_size )
378 msg_Warn( p_demux, "ALSA read overrun (%zu > %zu)",
379 i_correction_delta, p_sys->i_max_frame_size );
380 i_correction_delta = p_sys->i_max_frame_size;
381 snd_pcm_prepare( p_sys->p_alsa_pcm );
383 i_correct += i_correction_delta;
387 /* delay failed so reset */
388 msg_Warn( p_demux, "ALSA snd_pcm_delay failed (%s)", snd_strerror( i_err ) );
389 snd_pcm_prepare( p_sys->p_alsa_pcm );
393 p_block->i_pts = p_block->i_dts =
394 mdate() - INT64_C(1000000) * (mtime_t)i_correct /
395 2 / ( p_sys->b_stereo ? 2 : 1) / p_sys->i_sample_rate;
400 /*****************************************************************************
401 * OpenAudioDev: open and set up the audio device and probe for capabilities
402 *****************************************************************************/
403 static int OpenAudioDevAlsa( demux_t *p_demux )
405 demux_sys_t *p_sys = p_demux->p_sys;
406 const char *psz_device = p_sys->psz_device;
407 p_sys->p_alsa_pcm = NULL;
408 snd_pcm_hw_params_t *p_hw_params = NULL;
409 snd_pcm_uframes_t buffer_size;
410 snd_pcm_uframes_t chunk_size;
415 if( ( i_err = snd_pcm_open( &p_sys->p_alsa_pcm, psz_device,
416 SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0)
418 msg_Err( p_demux, "Cannot open ALSA audio device %s (%s)",
419 psz_device, snd_strerror( i_err ) );
423 if( ( i_err = snd_pcm_nonblock( p_sys->p_alsa_pcm, 1 ) ) < 0)
425 msg_Err( p_demux, "Cannot set ALSA nonblock (%s)",
426 snd_strerror( i_err ) );
430 /* Begin setting hardware parameters */
432 if( ( i_err = snd_pcm_hw_params_malloc( &p_hw_params ) ) < 0 )
435 "ALSA: cannot allocate hardware parameter structure (%s)",
436 snd_strerror( i_err ) );
440 if( ( i_err = snd_pcm_hw_params_any( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
443 "ALSA: cannot initialize hardware parameter structure (%s)",
444 snd_strerror( i_err ) );
448 /* Set Interleaved access */
449 if( ( i_err = snd_pcm_hw_params_set_access( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
451 msg_Err( p_demux, "ALSA: cannot set access type (%s)",
452 snd_strerror( i_err ) );
456 /* Set 16 bit little endian */
457 if( ( i_err = snd_pcm_hw_params_set_format( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_FORMAT_S16_LE ) ) < 0 )
459 msg_Err( p_demux, "ALSA: cannot set sample format (%s)",
460 snd_strerror( i_err ) );
464 /* Set sample rate */
465 #ifdef HAVE_ALSA_NEW_API
466 i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, &p_sys->i_sample_rate, NULL );
468 i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, p_sys->i_sample_rate, NULL );
472 msg_Err( p_demux, "ALSA: cannot set sample rate (%s)",
473 snd_strerror( i_err ) );
478 unsigned int channels = p_sys->b_stereo ? 2 : 1;
479 if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 )
481 channels = ( channels==1 ) ? 2 : 1;
482 msg_Warn( p_demux, "ALSA: cannot set channel count (%s). "
483 "Trying with channels=%d",
484 snd_strerror( i_err ),
486 if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 )
488 msg_Err( p_demux, "ALSA: cannot set channel count (%s)",
489 snd_strerror( i_err ) );
492 p_sys->b_stereo = ( channels == 2 );
495 /* Set metrics for buffer calculations later */
496 unsigned int buffer_time;
497 if( ( i_err = snd_pcm_hw_params_get_buffer_time_max(p_hw_params, &buffer_time, 0) ) < 0 )
499 msg_Err( p_demux, "ALSA: cannot get buffer time max (%s)",
500 snd_strerror( i_err ) );
503 if( buffer_time > 500000 ) buffer_time = 500000;
505 /* Set period time */
506 unsigned int period_time = buffer_time / 4;
507 #ifdef HAVE_ALSA_NEW_API
508 i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, &period_time, 0 );
510 i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, period_time, 0 );
514 msg_Err( p_demux, "ALSA: cannot set period time (%s)",
515 snd_strerror( i_err ) );
519 /* Set buffer time */
520 #ifdef HAVE_ALSA_NEW_API
521 i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, &buffer_time, 0 );
523 i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, buffer_time, 0 );
527 msg_Err( p_demux, "ALSA: cannot set buffer time (%s)",
528 snd_strerror( i_err ) );
532 /* Apply new hardware parameters */
533 if( ( i_err = snd_pcm_hw_params( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
535 msg_Err( p_demux, "ALSA: cannot set hw parameters (%s)",
536 snd_strerror( i_err ) );
540 /* Get various buffer metrics */
541 snd_pcm_hw_params_get_period_size( p_hw_params, &chunk_size, 0 );
542 snd_pcm_hw_params_get_buffer_size( p_hw_params, &buffer_size );
543 if( chunk_size == buffer_size )
546 "ALSA: period cannot equal buffer size (%lu == %lu)",
547 chunk_size, buffer_size);
551 int bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
552 int bits_per_frame = bits_per_sample * channels;
554 p_sys->i_alsa_chunk_size = chunk_size;
555 p_sys->i_alsa_frame_size = bits_per_frame / 8;
556 p_sys->i_max_frame_size = chunk_size * bits_per_frame / 8;
558 snd_pcm_hw_params_free( p_hw_params );
562 if( ( i_err = snd_pcm_prepare( p_sys->p_alsa_pcm ) ) < 0 )
565 "ALSA: cannot prepare audio interface for use (%s)",
566 snd_strerror( i_err ) );
570 snd_pcm_start( p_sys->p_alsa_pcm );
576 if( p_hw_params ) snd_pcm_hw_params_free( p_hw_params );
577 if( p_sys->p_alsa_pcm ) snd_pcm_close( p_sys->p_alsa_pcm );
578 p_sys->p_alsa_pcm = NULL;
584 static int OpenAudioDev( demux_t *p_demux )
586 demux_sys_t *p_sys = p_demux->p_sys;
587 if( OpenAudioDevAlsa( p_demux ) != VLC_SUCCESS )
590 msg_Dbg( p_demux, "opened adev=`%s' %s %dHz",
591 p_sys->psz_device, p_sys->b_stereo ? "stereo" : "mono",
592 p_sys->i_sample_rate );
595 es_format_Init( &fmt, AUDIO_ES, VLC_FOURCC('a','r','a','w') );
597 fmt.audio.i_channels = p_sys->b_stereo ? 2 : 1;
598 fmt.audio.i_rate = p_sys->i_sample_rate;
599 fmt.audio.i_bitspersample = 16;
600 fmt.audio.i_blockalign = fmt.audio.i_channels * fmt.audio.i_bitspersample / 8;
601 fmt.i_bitrate = fmt.audio.i_channels * fmt.audio.i_rate * fmt.audio.i_bitspersample;
603 msg_Dbg( p_demux, "new audio es %d channels %dHz",
604 fmt.audio.i_channels, fmt.audio.i_rate );
606 p_sys->p_es = es_out_Add( p_demux->out, &fmt );
611 /*****************************************************************************
612 * ProbeAudioDevAlsa: probe audio for capabilities
613 *****************************************************************************/
614 static bool ProbeAudioDevAlsa( demux_t *p_demux, const char *psz_device )
617 snd_pcm_t *p_alsa_pcm;
619 if( ( i_err = snd_pcm_open( &p_alsa_pcm, psz_device, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0 )
621 msg_Err( p_demux, "cannot open device %s for ALSA audio (%s)", psz_device, snd_strerror( i_err ) );
625 snd_pcm_close( p_alsa_pcm );