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( vlc_object_t *, demux_sys_t * );
115 static bool ProbeAudioDevAlsa( vlc_object_t *, const char *psz_device );
119 const char *psz_device; /* Alsa device from MRL */
124 unsigned int i_sample_rate;
126 size_t i_audio_max_frame_size;
127 block_t *p_block_audio;
128 es_out_id_t *p_es_audio;
133 snd_pcm_t *p_alsa_pcm;
134 size_t i_alsa_frame_size;
135 int i_alsa_chunk_size;
138 static int FindMainDevice( vlc_object_t *p_this, demux_sys_t *p_sys )
140 msg_Dbg( p_this, "opening device '%s'", p_sys->psz_device );
141 if( ProbeAudioDevAlsa( p_this, p_sys->psz_device ) )
143 msg_Dbg( p_this, "'%s' is an audio device", p_sys->psz_device );
144 p_sys->i_fd_audio = OpenAudioDev( p_this, p_sys );
147 if( p_sys->i_fd_audio < 0 )
152 /*****************************************************************************
153 * DemuxOpen: opens alsa device, access_demux callback
154 *****************************************************************************
156 * url: <alsa device>::::
158 *****************************************************************************/
159 static int DemuxOpen( vlc_object_t *p_this )
161 demux_t *p_demux = (demux_t*)p_this;
164 /* Only when selected */
165 if( *p_demux->psz_access == '\0' ) return VLC_EGENERIC;
168 p_demux->pf_control = DemuxControl;
169 p_demux->pf_demux = Demux;
170 p_demux->info.i_update = 0;
171 p_demux->info.i_title = 0;
172 p_demux->info.i_seekpoint = 0;
174 p_demux->p_sys = p_sys = calloc( 1, sizeof( demux_sys_t ) );
175 if( p_sys == NULL ) return VLC_ENOMEM;
177 p_sys->i_sample_rate = var_CreateGetInteger( p_demux, CFG_PREFIX "samplerate" );
178 p_sys->b_stereo = var_CreateGetBool( p_demux, CFG_PREFIX "stereo" );
179 p_sys->i_pts = var_CreateGetInteger( p_demux, CFG_PREFIX "caching" );
180 p_sys->psz_device = NULL;
181 p_sys->i_fd_audio = -1;
182 p_sys->p_es_audio = NULL;
183 p_sys->p_block_audio = NULL;
185 if( p_demux->psz_path && *p_demux->psz_path )
186 p_sys->psz_device = p_demux->psz_path;
188 p_sys->psz_device = ALSA_DEFAULT;
189 msg_Err( p_this, "Device is %s", p_sys->psz_device );
191 if( FindMainDevice( p_this, p_sys ) != 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 );
211 p_sys->i_fd_audio = -1;
213 if( p_sys->i_fd_audio >= 0 ) close( p_sys->i_fd_audio );
215 if( p_sys->p_block_audio ) block_Release( p_sys->p_block_audio );
219 /*****************************************************************************
221 *****************************************************************************/
222 static int DemuxControl( demux_t *p_demux, int i_query, va_list args )
224 demux_sys_t *p_sys = p_demux->p_sys;
230 /* Special for access_demux */
231 case DEMUX_CAN_PAUSE:
233 case DEMUX_SET_PAUSE_STATE:
234 case DEMUX_CAN_CONTROL_PACE:
235 pb = (bool*)va_arg( args, bool * );
239 case DEMUX_GET_PTS_DELAY:
240 pi64 = (int64_t*)va_arg( args, int64_t * );
241 *pi64 = (int64_t)p_sys->i_pts * 1000;
245 pi64 = (int64_t*)va_arg( args, int64_t * );
249 /* TODO implement others */
257 /*****************************************************************************
258 * Demux: Processes the audio frame
259 *****************************************************************************/
260 static int Demux( demux_t *p_demux )
262 demux_sys_t *p_sys = p_demux->p_sys;
265 fd.fd = p_sys->i_fd_audio;
266 fd.events = POLLIN|POLLPRI;
270 if( poll( &fd, 1, 500 ) ) /* Timeout after 0.5 seconds since I don't know if pf_demux can be blocking. */
272 if( fd.revents & (POLLIN|POLLPRI) )
274 block_t *p_block = GrabAudio( p_demux );
277 es_out_Control( p_demux->out, ES_OUT_SET_PCR, p_block->i_pts );
278 es_out_Send( p_demux->out, p_sys->p_es_audio, p_block );
287 /*****************************************************************************
288 * GrabAudio: Grab an audio frame
289 *****************************************************************************/
290 static block_t* GrabAudio( demux_t *p_demux )
292 demux_sys_t *p_sys = p_demux->p_sys;
293 int i_read = 0, i_correct;
296 printf("%s %d\n",__func__,__LINE__);
297 if( p_sys->p_block_audio ) p_block = p_sys->p_block_audio;
298 else p_block = block_New( p_demux, p_sys->i_audio_max_frame_size );
302 msg_Warn( p_demux, "cannot get block" );
306 p_sys->p_block_audio = p_block;
309 i_read = snd_pcm_readi( p_sys->p_alsa_pcm, p_block->p_buffer, p_sys->i_alsa_chunk_size );
319 snd_pcm_prepare( p_sys->p_alsa_pcm );
323 i_resume = snd_pcm_resume( p_sys->p_alsa_pcm );
324 if( i_resume < 0 && i_resume != -EAGAIN ) snd_pcm_prepare( p_sys->p_alsa_pcm );
327 msg_Err( p_demux, "Failed to read alsa frame (%s)", snd_strerror( i_read ) );
333 /* convert from frames to bytes */
334 i_read *= p_sys->i_alsa_frame_size;
337 if( i_read <= 0 ) return 0;
339 p_block->i_buffer = i_read;
340 p_sys->p_block_audio = 0;
342 /* Correct the date because of kernel buffering */
346 snd_pcm_sframes_t delay = 0;
347 if( ( i_err = snd_pcm_delay( p_sys->p_alsa_pcm, &delay ) ) >= 0 )
349 size_t i_correction_delta = delay * p_sys->i_alsa_frame_size;
350 /* Test for overrun */
351 if( i_correction_delta > p_sys->i_audio_max_frame_size )
353 msg_Warn( p_demux, "ALSA read overrun (%zu > %zu)",
354 i_correction_delta, p_sys->i_audio_max_frame_size );
355 i_correction_delta = p_sys->i_audio_max_frame_size;
356 snd_pcm_prepare( p_sys->p_alsa_pcm );
358 i_correct += i_correction_delta;
362 /* delay failed so reset */
363 msg_Warn( p_demux, "ALSA snd_pcm_delay failed (%s)", snd_strerror( i_err ) );
364 snd_pcm_prepare( p_sys->p_alsa_pcm );
368 p_block->i_pts = p_block->i_dts =
369 mdate() - INT64_C(1000000) * (mtime_t)i_correct /
370 2 / ( p_sys->b_stereo ? 2 : 1) / p_sys->i_sample_rate;
375 /*****************************************************************************
376 * OpenAudioDev: open and set up the audio device and probe for capabilities
377 *****************************************************************************/
378 static int OpenAudioDevAlsa( vlc_object_t *p_this, demux_sys_t *p_sys )
380 const char *psz_device = p_sys->psz_device;
381 p_sys->p_alsa_pcm = NULL;
382 snd_pcm_hw_params_t *p_hw_params = NULL;
383 snd_pcm_uframes_t buffer_size;
384 snd_pcm_uframes_t chunk_size;
389 if( ( i_err = snd_pcm_open( &p_sys->p_alsa_pcm, psz_device,
390 SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0)
392 msg_Err( p_this, "Cannot open ALSA audio device %s (%s)",
393 psz_device, snd_strerror( i_err ) );
397 if( ( i_err = snd_pcm_nonblock( p_sys->p_alsa_pcm, 1 ) ) < 0)
399 msg_Err( p_this, "Cannot set ALSA nonblock (%s)",
400 snd_strerror( i_err ) );
404 /* Begin setting hardware parameters */
406 if( ( i_err = snd_pcm_hw_params_malloc( &p_hw_params ) ) < 0 )
409 "ALSA: cannot allocate hardware parameter structure (%s)",
410 snd_strerror( i_err ) );
414 if( ( i_err = snd_pcm_hw_params_any( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
417 "ALSA: cannot initialize hardware parameter structure (%s)",
418 snd_strerror( i_err ) );
422 /* Set Interleaved access */
423 if( ( i_err = snd_pcm_hw_params_set_access( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
425 msg_Err( p_this, "ALSA: cannot set access type (%s)",
426 snd_strerror( i_err ) );
430 /* Set 16 bit little endian */
431 if( ( i_err = snd_pcm_hw_params_set_format( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_FORMAT_S16_LE ) ) < 0 )
433 msg_Err( p_this, "ALSA: cannot set sample format (%s)",
434 snd_strerror( i_err ) );
438 /* Set sample rate */
439 #ifdef HAVE_ALSA_NEW_API
440 i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, &p_sys->i_sample_rate, NULL );
442 i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, p_sys->i_sample_rate, NULL );
446 msg_Err( p_this, "ALSA: cannot set sample rate (%s)",
447 snd_strerror( i_err ) );
452 unsigned int channels = p_sys->b_stereo ? 2 : 1;
453 if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 )
455 channels = ( channels==1 ) ? 2 : 1;
456 msg_Warn( p_this, "ALSA: cannot set channel count (%s). "
457 "Trying with channels=%d",
458 snd_strerror( i_err ),
460 if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 )
462 msg_Err( p_this, "ALSA: cannot set channel count (%s)",
463 snd_strerror( i_err ) );
466 p_sys->b_stereo = ( channels == 2 );
469 /* Set metrics for buffer calculations later */
470 unsigned int buffer_time;
471 if( ( i_err = snd_pcm_hw_params_get_buffer_time_max(p_hw_params, &buffer_time, 0) ) < 0 )
473 msg_Err( p_this, "ALSA: cannot get buffer time max (%s)",
474 snd_strerror( i_err ) );
477 if( buffer_time > 500000 ) buffer_time = 500000;
479 /* Set period time */
480 unsigned int period_time = buffer_time / 4;
481 #ifdef HAVE_ALSA_NEW_API
482 i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, &period_time, 0 );
484 i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, period_time, 0 );
488 msg_Err( p_this, "ALSA: cannot set period time (%s)",
489 snd_strerror( i_err ) );
493 /* Set buffer time */
494 #ifdef HAVE_ALSA_NEW_API
495 i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, &buffer_time, 0 );
497 i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, buffer_time, 0 );
501 msg_Err( p_this, "ALSA: cannot set buffer time (%s)",
502 snd_strerror( i_err ) );
506 /* Apply new hardware parameters */
507 if( ( i_err = snd_pcm_hw_params( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
509 msg_Err( p_this, "ALSA: cannot set hw parameters (%s)",
510 snd_strerror( i_err ) );
514 /* Get various buffer metrics */
515 snd_pcm_hw_params_get_period_size( p_hw_params, &chunk_size, 0 );
516 snd_pcm_hw_params_get_buffer_size( p_hw_params, &buffer_size );
517 if( chunk_size == buffer_size )
520 "ALSA: period cannot equal buffer size (%lu == %lu)",
521 chunk_size, buffer_size);
525 int bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
526 int bits_per_frame = bits_per_sample * channels;
528 p_sys->i_alsa_chunk_size = chunk_size;
529 p_sys->i_alsa_frame_size = bits_per_frame / 8;
530 p_sys->i_audio_max_frame_size = chunk_size * bits_per_frame / 8;
532 snd_pcm_hw_params_free( p_hw_params );
536 if( ( i_err = snd_pcm_prepare( p_sys->p_alsa_pcm ) ) < 0 )
539 "ALSA: cannot prepare audio interface for use (%s)",
540 snd_strerror( i_err ) );
544 if( !p_sys->psz_device )
545 p_sys->psz_device = strdup( ALSA_DEFAULT );
547 /* Return a fake handle so other tests work */
552 if( p_hw_params ) snd_pcm_hw_params_free( p_hw_params );
553 if( p_sys->p_alsa_pcm ) snd_pcm_close( p_sys->p_alsa_pcm );
559 static int OpenAudioDev( vlc_object_t *p_this, demux_sys_t *p_sys )
561 int i_fd = OpenAudioDevAlsa( p_this, p_sys );
566 msg_Dbg( p_this, "opened adev=`%s' %s %dHz",
567 p_sys->psz_device, p_sys->b_stereo ? "stereo" : "mono",
568 p_sys->i_sample_rate );
571 es_format_Init( &fmt, AUDIO_ES, VLC_FOURCC('a','r','a','w') );
573 fmt.audio.i_channels = p_sys->b_stereo ? 2 : 1;
574 fmt.audio.i_rate = p_sys->i_sample_rate;
575 fmt.audio.i_bitspersample = 16;
576 fmt.audio.i_blockalign = fmt.audio.i_channels * fmt.audio.i_bitspersample / 8;
577 fmt.i_bitrate = fmt.audio.i_channels * fmt.audio.i_rate * fmt.audio.i_bitspersample;
579 msg_Dbg( p_this, "new audio es %d channels %dHz",
580 fmt.audio.i_channels, fmt.audio.i_rate );
582 demux_t *p_demux = (demux_t *)p_this;
583 p_sys->p_es_audio = es_out_Add( p_demux->out, &fmt );
588 /*****************************************************************************
589 * ProbeAudioDevAlsa: probe audio for capabilities
590 *****************************************************************************/
591 static bool ProbeAudioDevAlsa( vlc_object_t *p_this, const char *psz_device )
594 snd_pcm_t *p_alsa_pcm;
596 if( ( i_err = snd_pcm_open( &p_alsa_pcm, psz_device, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0 )
598 msg_Err( p_this, "cannot open device %s for ALSA audio (%s)", psz_device, snd_strerror( i_err ) );
602 snd_pcm_close( p_alsa_pcm );