/*****************************************************************************
- * alsa.c : Alsa input module for vlc
+ * alsa.c: ALSA capture module for VLC
*****************************************************************************
- * Copyright (C) 2002-2009 the VideoLAN team
- * $Id$
+ * Copyright (C) 2012 RĂ©mi Denis-Courmont
*
- * Authors: Benjamin Pracht <bigben at videolan dot org>
- * Richard Hosking <richard at hovis dot net>
- * Antoine Cellerier <dionoea at videolan d.t org>
- * Dennis Lou <dlou99 at yahoo dot com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 2.1 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
*
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
- *****************************************************************************/
-
-/*
- * ALSA support based on parts of
- * http://www.equalarea.com/paul/alsa-audio.html
- * and hints taken from alsa-utils (aplay/arecord)
- * http://www.alsa-project.org
- */
-
-/*****************************************************************************
- * Preamble
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
#ifdef HAVE_CONFIG_H
-# include "config.h"
+# include <config.h>
#endif
-#include <vlc_common.h>
-#include <vlc_plugin.h>
-#include <vlc_access.h>
-#include <vlc_demux.h>
-#include <vlc_input.h>
-
-#include <ctype.h>
-#include <fcntl.h>
-#include <unistd.h>
-#include <sys/ioctl.h>
-#include <sys/mman.h>
-
-#include <sys/soundcard.h>
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-#include <alsa/asoundlib.h>
-
+#include <assert.h>
+#include <sys/types.h>
#include <poll.h>
+#include <alsa/asoundlib.h>
-/*****************************************************************************
- * Module descriptior
- *****************************************************************************/
-
-static int DemuxOpen ( vlc_object_t * );
-static void DemuxClose( vlc_object_t * );
+#include <vlc_common.h>
+#include <vlc_demux.h>
+#include <vlc_aout.h>
+#include <vlc_plugin.h>
-#define STEREO_TEXT N_( "Stereo" )
-#define STEREO_LONGTEXT N_( \
- "Capture the audio stream in stereo." )
+#define HELP_TEXT N_( \
+ "Pass alsa:// to open the default ALSA capture device, " \
+ "or alsa://SOURCE to open a specific device named SOURCE.")
+#define STEREO_TEXT N_("Stereo")
+#define RATE_TEXT N_("Sample rate")
-#define SAMPLERATE_TEXT N_( "Samplerate" )
-#define SAMPLERATE_LONGTEXT N_( \
- "Samplerate of the captured audio stream, in Hz (eg: 11025, 22050, 44100, 48000)" )
+static int Open (vlc_object_t *);
+static void Close (vlc_object_t *);
-#define CACHING_TEXT N_("Caching value in ms")
-#define CACHING_LONGTEXT N_( \
- "Caching value for Alsa captures. This " \
- "value should be set in milliseconds." )
+static const int rate_values[] = { 192000, 176400,
+ 96000, 88200, 48000, 44100,
+ 32000, 22050, 24000, 16000,
+ 11025, 8000, 4000
+};
+static const char *const rate_names[] = { N_("192000 Hz"), N_("176400 Hz"),
+ N_("96000 Hz"), N_("88200 Hz"), N_("48000 Hz"), N_("44100 Hz"),
+ N_("32000 Hz"), N_("22050 Hz"), N_("24000 Hz"), N_("16000 Hz"),
+ N_("11025 Hz"), N_("8000 Hz"), N_("4000 Hz")
+};
-#define ALSA_DEFAULT "hw"
-#define CFG_PREFIX "alsa-"
+vlc_module_begin ()
+ set_shortname (N_("ALSA"))
+ set_description (N_("ALSA audio capture"))
+ set_capability ("access_demux", 0)
+ set_category (CAT_INPUT)
+ set_subcategory (SUBCAT_INPUT_ACCESS)
+ set_help (HELP_TEXT)
+
+ add_obsolete_string ("alsa-format") /* since 2.1.0 */
+ add_bool ("alsa-stereo", true, STEREO_TEXT, STEREO_TEXT, true)
+ add_integer ("alsa-samplerate", 48000, RATE_TEXT, RATE_TEXT, true)
+ change_integer_list (rate_values, rate_names)
+
+ add_shortcut ("alsa")
+ set_callbacks (Open, Close)
+vlc_module_end ()
+
+/** Helper for ALSA -> VLC debugging output */
+/** XXX: duplicated from ALSA 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;
-vlc_module_begin()
- set_shortname( N_("Alsa") )
- set_description( N_("Alsa audio capture input") )
- set_category( CAT_INPUT )
- set_subcategory( SUBCAT_INPUT_ACCESS )
+ if (unlikely(snd_output_buffer_open (&output)))
+ return;
- add_shortcut( "alsa" )
- set_capability( "access_demux", 10 )
- set_callbacks( DemuxOpen, DemuxClose )
+ int val = cb (p, output);
+ if (val)
+ {
+ msg_Warn (obj, "cannot get info: %s", snd_strerror (val));
+ return;
+ }
- add_bool( CFG_PREFIX "stereo", true, NULL, STEREO_TEXT, STEREO_LONGTEXT,
- true )
- add_integer( CFG_PREFIX "samplerate", 48000, NULL, SAMPLERATE_TEXT,
- SAMPLERATE_LONGTEXT, true )
- add_integer( CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL,
- CACHING_TEXT, CACHING_LONGTEXT, true )
-vlc_module_end()
+ 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)
-/*****************************************************************************
- * Access: local prototypes
- *****************************************************************************/
+static void DumpDevice (vlc_object_t *obj, snd_pcm_t *pcm)
+{
+ snd_pcm_info_t *info;
-static int DemuxControl( demux_t *, int, va_list );
+ 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 int Demux( demux_t * );
+static void DumpDeviceStatus (vlc_object_t *obj, snd_pcm_t *pcm)
+{
+ snd_pcm_status_t *status;
-static block_t* GrabAudio( demux_t *p_demux );
+ 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)
-static int OpenAudioDev( demux_t * );
-static bool ProbeAudioDevAlsa( demux_t *, const char *psz_device );
struct demux_sys_t
{
- const char *psz_device; /* Alsa device from MRL */
-
- /* Audio */
- int i_cache;
- unsigned int i_sample_rate;
- bool b_stereo;
- size_t i_max_frame_size;
- block_t *p_block;
- es_out_id_t *p_es;
-
- /* ALSA Audio */
- snd_pcm_t *p_alsa_pcm;
- size_t i_alsa_frame_size;
- int i_alsa_chunk_size;
-
- int64_t i_next_demux_date; /* Used to handle alsa:// as input-slave properly */
+ snd_pcm_t *pcm;
+ es_out_id_t *es;
+ vlc_thread_t thread;
+
+ mtime_t start;
+ mtime_t caching;
+ snd_pcm_uframes_t period_size;
+ unsigned rate;
};
-static int FindMainDevice( demux_t *p_demux )
+static void Poll (snd_pcm_t *pcm, int canc)
{
- /* TODO: if using default device, loop through all alsa devices until
- * one works. */
- msg_Dbg( p_demux, "opening device '%s'", p_demux->p_sys->psz_device );
- if( ProbeAudioDevAlsa( p_demux, p_demux->p_sys->psz_device ) )
+ int n = snd_pcm_poll_descriptors_count (pcm);
+ struct pollfd ufd[n];
+ unsigned short revents;
+
+ snd_pcm_poll_descriptors (pcm, ufd, n);
+ do
{
- msg_Dbg( p_demux, "'%s' is an audio device",
- p_demux->p_sys->psz_device );
- OpenAudioDev( p_demux );
+ vlc_restorecancel (canc);
+ while (poll (ufd, n, -1) == -1);
+ canc = vlc_savecancel ();
+ snd_pcm_poll_descriptors_revents (pcm, ufd, n, &revents);
}
-
- if( p_demux->p_sys->p_alsa_pcm == NULL )
- return VLC_EGENERIC;
- return VLC_SUCCESS;
+ while (!revents);
}
-static void ListAvailableDevices( demux_t *p_demux )
+static void *Thread (void *data)
{
- snd_ctl_card_info_t *p_info = NULL;
- snd_ctl_card_info_alloca( &p_info );
-
- snd_pcm_info_t *p_pcminfo = NULL;
- snd_pcm_info_alloca( &p_pcminfo );
+ demux_t *demux = data;
+ demux_sys_t *sys = demux->p_sys;
+ snd_pcm_t *pcm = sys->pcm;
+ size_t bytes;
+ int canc, val;
+
+ canc = vlc_savecancel ();
+ bytes = snd_pcm_frames_to_bytes (pcm, sys->period_size);
+ val = snd_pcm_start (pcm);
+ if (val)
+ {
+ msg_Err (demux, "cannot prepare device: %s", snd_strerror (val));
+ return NULL;
+ }
- msg_Dbg( p_demux, "Available alsa capture devices:" );
- int i_card = -1;
- while( !snd_card_next( &i_card ) && i_card >= 0 )
+ for (;;)
{
- char psz_devname[10];
- snprintf( psz_devname, 10, "hw:%d", i_card );
+ block_t *block = block_Alloc (bytes);
+ if (unlikely(block == NULL))
+ break;
- snd_ctl_t *p_ctl = NULL;
- if( snd_ctl_open( &p_ctl, psz_devname, 0 ) < 0 ) continue;
+ /* Wait for data */
+ Poll (pcm, canc);
- snd_ctl_card_info( p_ctl, p_info );
- msg_Dbg( p_demux, " %s (%s)",
- snd_ctl_card_info_get_id( p_info ),
- snd_ctl_card_info_get_name( p_info ) );
+ /* Read data */
+ snd_pcm_sframes_t frames, delay;
+ mtime_t pts;
- int i_dev = -1;
- while( !snd_ctl_pcm_next_device( p_ctl, &i_dev ) && i_dev >= 0 )
+ frames = snd_pcm_readi (pcm, block->p_buffer, sys->period_size);
+ pts = mdate ();
+ if (frames < 0)
{
- snd_pcm_info_set_device( p_pcminfo, i_dev );
- snd_pcm_info_set_subdevice( p_pcminfo, 0 );
- snd_pcm_info_set_stream( p_pcminfo, SND_PCM_STREAM_CAPTURE );
- if( snd_ctl_pcm_info( p_ctl, p_pcminfo ) < 0 ) continue;
-
- msg_Dbg( p_demux, " hw:%d,%d : %s (%s)", i_card, i_dev,
- snd_pcm_info_get_id( p_pcminfo ),
- snd_pcm_info_get_name( p_pcminfo ) );
+ block_Release (block);
+ if (frames == -EAGAIN)
+ continue;
+
+ val = snd_pcm_recover (pcm, frames, 1);
+ if (val == 0)
+ {
+ msg_Warn (demux, "cannot read samples: %s",
+ snd_strerror (frames));
+ snd_pcm_state_t state = snd_pcm_state (pcm);
+ switch (state)
+ {
+ case SND_PCM_STATE_PREPARED:
+ val = snd_pcm_start (pcm);
+ if (val < 0)
+ {
+ msg_Err (demux, "cannot prepare device: %s",
+ snd_strerror (val));
+ return NULL;
+ }
+ continue;
+ case SND_PCM_STATE_RUNNING:
+ continue;
+ default:
+ break;
+ }
+ }
+ msg_Err (demux, "cannot recover record stream: %s",
+ snd_strerror (val));
+ DumpDeviceStatus (demux, pcm);
+ break;
}
- snd_ctl_close( p_ctl );
- }
-}
+ /* Compute time stamp */
+ if (snd_pcm_delay (pcm, &delay))
+ delay = 0;
+ delay += frames;
+ pts -= (CLOCK_FREQ * delay) / sys->rate;
-/*****************************************************************************
- * DemuxOpen: opens alsa device, access_demux callback
- *****************************************************************************
- *
- * url: <alsa device>::::
- *
- *****************************************************************************/
-static int DemuxOpen( vlc_object_t *p_this )
-{
- demux_t *p_demux = (demux_t*)p_this;
- demux_sys_t *p_sys;
-
- /* Only when selected */
- if( *p_demux->psz_access == '\0' ) return VLC_EGENERIC;
-
- /* Set up p_demux */
- p_demux->pf_control = DemuxControl;
- p_demux->pf_demux = Demux;
- p_demux->info.i_update = 0;
- p_demux->info.i_title = 0;
- p_demux->info.i_seekpoint = 0;
-
- p_demux->p_sys = p_sys = calloc( 1, sizeof( demux_sys_t ) );
- if( p_sys == NULL ) return VLC_ENOMEM;
-
- p_sys->i_sample_rate = var_CreateGetInteger( p_demux, CFG_PREFIX "samplerate" );
- p_sys->b_stereo = var_CreateGetBool( p_demux, CFG_PREFIX "stereo" );
- p_sys->i_cache = var_CreateGetInteger( p_demux, CFG_PREFIX "caching" );
- p_sys->p_es = NULL;
- p_sys->p_block = NULL;
- p_sys->i_next_demux_date = -1;
-
- if( p_demux->psz_path && *p_demux->psz_path )
- p_sys->psz_device = p_demux->psz_path;
- else
- {
- p_sys->psz_device = ALSA_DEFAULT;
- ListAvailableDevices( p_demux );
- }
+ block->i_buffer = snd_pcm_frames_to_bytes (pcm, frames);
+ block->i_nb_samples = frames;
+ block->i_pts = pts;
+ block->i_length = (CLOCK_FREQ * frames) / sys->rate;
- if( FindMainDevice( p_demux ) != VLC_SUCCESS )
- {
- DemuxClose( p_this );
- return VLC_EGENERIC;
+ es_out_Control (demux->out, ES_OUT_SET_PCR, block->i_pts);
+ es_out_Send (demux->out, sys->es, block);
}
-
- return VLC_SUCCESS;
+ return NULL;
}
-/*****************************************************************************
- * Close: close device, free resources
- *****************************************************************************/
-static void DemuxClose( vlc_object_t *p_this )
+static int Control (demux_t *demux, int query, va_list ap)
{
- demux_t *p_demux = (demux_t *)p_this;
- demux_sys_t *p_sys = p_demux->p_sys;
+ demux_sys_t *sys = demux->p_sys;
- if( p_sys->p_alsa_pcm )
+ switch (query)
{
- snd_pcm_close( p_sys->p_alsa_pcm );
- }
-
- if( p_sys->p_block ) block_Release( p_sys->p_block );
+ case DEMUX_GET_TIME:
+ *va_arg (ap, int64_t *) = mdate () - sys->start;
+ break;
- free( p_sys );
-}
+ case DEMUX_GET_PTS_DELAY:
+ *va_arg (ap, int64_t *) = sys->caching;
+ break;
-/*****************************************************************************
- * DemuxControl:
- *****************************************************************************/
-static int DemuxControl( demux_t *p_demux, int i_query, va_list args )
-{
- demux_sys_t *p_sys = p_demux->p_sys;
- bool *pb;
- int64_t *pi64;
+ //case DEMUX_SET_NEXT_DEMUX_TIME: still needed?
- switch( i_query )
- {
- /* Special for access_demux */
+ case DEMUX_HAS_UNSUPPORTED_META:
+ case DEMUX_CAN_RECORD:
case DEMUX_CAN_PAUSE:
- case DEMUX_CAN_SEEK:
- case DEMUX_SET_PAUSE_STATE:
case DEMUX_CAN_CONTROL_PACE:
- pb = (bool*)va_arg( args, bool * );
- *pb = false;
- return VLC_SUCCESS;
-
- case DEMUX_GET_PTS_DELAY:
- pi64 = (int64_t*)va_arg( args, int64_t * );
- *pi64 = (int64_t)p_sys->i_cache * 1000;
- return VLC_SUCCESS;
-
- case DEMUX_GET_TIME:
- pi64 = (int64_t*)va_arg( args, int64_t * );
- *pi64 = mdate();
- return VLC_SUCCESS;
-
- case DEMUX_SET_NEXT_DEMUX_TIME:
- p_sys->i_next_demux_date = (int64_t)va_arg( args, int64_t );
- return VLC_SUCCESS;
+ case DEMUX_CAN_CONTROL_RATE:
+ case DEMUX_CAN_SEEK:
+ *va_arg (ap, bool *) = false;
+ break;;
- /* TODO implement others */
default:
return VLC_EGENERIC;
}
- return VLC_EGENERIC;
+ return VLC_SUCCESS;
}
-/*****************************************************************************
- * Demux: Processes the audio frame
- *****************************************************************************/
-static int Demux( demux_t *p_demux )
-{
- demux_sys_t *p_sys = p_demux->p_sys;
-
- block_t *p_block = NULL;
-
- do
- {
- if( p_block )
- {
- es_out_Send( p_demux->out, p_sys->p_es, p_block );
- p_block = NULL;
- }
-
- /* Wait for data */
- int i_wait = snd_pcm_wait( p_sys->p_alsa_pcm, 500 );
- switch( i_wait )
- {
- case 1:
- {
- p_block = GrabAudio( p_demux );
- if( p_block )
- es_out_Control( p_demux->out, ES_OUT_SET_PCR, p_block->i_pts );
- }
-
- /* FIXME: this is a copy paste from below. Shouldn't be needed
- * twice. */
- case -EPIPE:
- /* xrun */
- snd_pcm_prepare( p_sys->p_alsa_pcm );
- break;
- case -ESTRPIPE:
- {
- /* suspend */
- int i_resume = snd_pcm_resume( p_sys->p_alsa_pcm );
- if( i_resume < 0 && i_resume != -EAGAIN ) snd_pcm_prepare( p_sys->p_alsa_pcm );
- break;
- }
- /* </FIXME> */
- }
- } while( p_block && p_sys->i_next_demux_date > 0 &&
- p_block->i_pts < p_sys->i_next_demux_date );
+static const vlc_fourcc_t formats[] = {
+ [SND_PCM_FORMAT_S8] = VLC_CODEC_S8,
+ [SND_PCM_FORMAT_U8] = VLC_CODEC_U8,
+ [SND_PCM_FORMAT_S16_LE] = VLC_CODEC_S16L,
+ [SND_PCM_FORMAT_S16_BE] = VLC_CODEC_S16B,
+ [SND_PCM_FORMAT_U16_LE] = VLC_CODEC_U16L,
+ [SND_PCM_FORMAT_U16_BE] = VLC_CODEC_U16B,
+ [SND_PCM_FORMAT_S24_LE] = VLC_CODEC_S24L32,
+ [SND_PCM_FORMAT_S24_BE] = VLC_CODEC_S24B32,
+ [SND_PCM_FORMAT_U24_LE] = VLC_CODEC_U32L, // TODO: replay gain
+ [SND_PCM_FORMAT_U24_BE] = VLC_CODEC_U32B, // ^
+ [SND_PCM_FORMAT_S32_LE] = VLC_CODEC_S32L,
+ [SND_PCM_FORMAT_S32_BE] = VLC_CODEC_S32B,
+ [SND_PCM_FORMAT_U32_LE] = VLC_CODEC_U32L,
+ [SND_PCM_FORMAT_U32_BE] = VLC_CODEC_U32B,
+ [SND_PCM_FORMAT_FLOAT_LE] = VLC_CODEC_F32L,
+ [SND_PCM_FORMAT_FLOAT_BE] = VLC_CODEC_F32B,
+ [SND_PCM_FORMAT_FLOAT64_LE] = VLC_CODEC_F32L,
+ [SND_PCM_FORMAT_FLOAT64_BE] = VLC_CODEC_F32B,
+ //[SND_PCM_FORMAT_IEC958_SUBFRAME_LE] = VLC_CODEC_SPDIFL,
+ //[SND_PCM_FORMAT_IEC958_SUBFRAME_BE] = VLC_CODEC_SPDIFB,
+ [SND_PCM_FORMAT_MU_LAW] = VLC_CODEC_MULAW,
+ [SND_PCM_FORMAT_A_LAW] = VLC_CODEC_ALAW,
+ //[SND_PCM_FORMAT_IMA_ADPCM] = VLC_CODEC_ADPCM_?, // XXX: which one?
+ [SND_PCM_FORMAT_MPEG] = VLC_CODEC_MPGA,
+ [SND_PCM_FORMAT_GSM] = VLC_CODEC_GSM,
+ //[SND_PCM_FORMAT_SPECIAL] = VLC_CODEC_?
+ [SND_PCM_FORMAT_S24_3LE] = VLC_CODEC_S24L,
+ [SND_PCM_FORMAT_S24_3BE] = VLC_CODEC_S24B,
+ [SND_PCM_FORMAT_U24_3LE] = VLC_CODEC_U24L,
+ [SND_PCM_FORMAT_U24_3BE] = VLC_CODEC_U24B,
+ [SND_PCM_FORMAT_S20_3LE] = VLC_CODEC_S24L, // TODO: replay gain
+ [SND_PCM_FORMAT_S20_3BE] = VLC_CODEC_S24B, // ^
+ [SND_PCM_FORMAT_U20_3LE] = VLC_CODEC_U24L, // ^
+ [SND_PCM_FORMAT_U20_3BE] = VLC_CODEC_U24B, // ^
+ [SND_PCM_FORMAT_S18_3LE] = VLC_CODEC_S24L, // ^
+ [SND_PCM_FORMAT_S18_3BE] = VLC_CODEC_S24B, // ^
+ [SND_PCM_FORMAT_U18_3LE] = VLC_CODEC_U24L, // ^
+ [SND_PCM_FORMAT_U18_3BE] = VLC_CODEC_U24B, // ^
+};
- if( p_block )
- es_out_Send( p_demux->out, p_sys->p_es, p_block );
+#ifdef WORDS_BIGENDIAN
+# define C(f) f##BE, f##LE
+#else
+# define C(f) f##LE, f##BE
+#endif
- return 1;
-}
+/* Formats in order of decreasing preference */
+static const uint8_t choices[] = {
+ C(SND_PCM_FORMAT_FLOAT_),
+ C(SND_PCM_FORMAT_S32_),
+ C(SND_PCM_FORMAT_U32_),
+ C(SND_PCM_FORMAT_S16_),
+ C(SND_PCM_FORMAT_U16_),
+ C(SND_PCM_FORMAT_FLOAT64_),
+ C(SND_PCM_FORMAT_S24_3),
+ C(SND_PCM_FORMAT_U24_3),
+ SND_PCM_FORMAT_MPEG,
+ SND_PCM_FORMAT_GSM,
+ SND_PCM_FORMAT_MU_LAW,
+ SND_PCM_FORMAT_A_LAW,
+ SND_PCM_FORMAT_S8,
+ SND_PCM_FORMAT_U8,
+};
+static uint16_t channel_maps[] = {
+ AOUT_CHAN_CENTER, AOUT_CHANS_2_0, AOUT_CHANS_3_0 /* ? */,
+ AOUT_CHANS_4_0, AOUT_CHANS_5_0 /* ? */, AOUT_CHANS_5_1,
+ /* TODO: support 7-8 channels - need channels reodering */
+};
-/*****************************************************************************
- * GrabAudio: Grab an audio frame
- *****************************************************************************/
-static block_t* GrabAudio( demux_t *p_demux )
+static int Open (vlc_object_t *obj)
{
- demux_sys_t *p_sys = p_demux->p_sys;
- int i_read, i_correct;
- block_t *p_block;
-
- if( p_sys->p_block ) p_block = p_sys->p_block;
- else p_block = block_New( p_demux, p_sys->i_max_frame_size );
-
- if( !p_block )
- {
- msg_Warn( p_demux, "cannot get block" );
- return 0;
- }
-
- p_sys->p_block = p_block;
-
- /* ALSA */
- i_read = snd_pcm_readi( p_sys->p_alsa_pcm, p_block->p_buffer, p_sys->i_alsa_chunk_size );
- if( i_read <= 0 )
- {
- int i_resume;
- switch( i_read )
- {
- case -EAGAIN:
- break;
- case -EPIPE:
- /* xrun */
- snd_pcm_prepare( p_sys->p_alsa_pcm );
- break;
- case -ESTRPIPE:
- /* suspend */
- i_resume = snd_pcm_resume( p_sys->p_alsa_pcm );
- if( i_resume < 0 && i_resume != -EAGAIN ) snd_pcm_prepare( p_sys->p_alsa_pcm );
- break;
- default:
- msg_Err( p_demux, "Failed to read alsa frame (%s)", snd_strerror( i_read ) );
- return 0;
- }
- }
- else
- {
- /* convert from frames to bytes */
- i_read *= p_sys->i_alsa_frame_size;
- }
-
- if( i_read <= 0 ) return 0;
-
- p_block->i_buffer = i_read;
- p_sys->p_block = 0;
-
- /* Correct the date because of kernel buffering */
- i_correct = i_read;
- /* ALSA */
- int i_err;
- snd_pcm_sframes_t delay = 0;
- if( ( i_err = snd_pcm_delay( p_sys->p_alsa_pcm, &delay ) ) >= 0 )
+ demux_t *demux = (demux_t *)obj;
+ demux_sys_t *sys = malloc (sizeof (*sys));
+
+ if (unlikely(sys == NULL))
+ return VLC_ENOMEM;
+
+ /* Open the device */
+ const char *device = demux->psz_location;
+ if (device == NULL || !device[0])
+ device = "default";
+
+ const int mode = SND_PCM_NONBLOCK
+ /*| SND_PCM_NO_AUTO_RESAMPLE*/
+ | SND_PCM_NO_AUTO_CHANNELS
+ /*| SND_PCM_NO_AUTO_FORMAT*/;
+ snd_pcm_t *pcm;
+ int val = snd_pcm_open (&pcm, device, SND_PCM_STREAM_CAPTURE, mode);
+ if (val != 0)
{
- size_t i_correction_delta = delay * p_sys->i_alsa_frame_size;
- /* Test for overrun */
- if( i_correction_delta > p_sys->i_max_frame_size )
- {
- msg_Warn( p_demux, "ALSA read overrun (%zu > %zu)",
- i_correction_delta, p_sys->i_max_frame_size );
- i_correction_delta = p_sys->i_max_frame_size;
- snd_pcm_prepare( p_sys->p_alsa_pcm );
- }
- i_correct += i_correction_delta;
- }
- else
- {
- /* delay failed so reset */
- msg_Warn( p_demux, "ALSA snd_pcm_delay failed (%s)", snd_strerror( i_err ) );
- snd_pcm_prepare( p_sys->p_alsa_pcm );
+ msg_Err (demux, "cannot open ALSA device \"%s\": %s", device,
+ snd_strerror (val));
+ free (sys);
+ return VLC_EGENERIC;
}
+ sys->pcm = pcm;
+ msg_Dbg (demux, "using ALSA device: %s", device);
+ DumpDevice (VLC_OBJECT(demux), pcm);
- /* Timestamp */
- p_block->i_pts = p_block->i_dts =
- mdate() - INT64_C(1000000) * (mtime_t)i_correct /
- 2 / ( p_sys->b_stereo ? 2 : 1) / p_sys->i_sample_rate;
+ /* Negotiate capture parameters */
+ snd_pcm_hw_params_t *hw;
+ es_format_t fmt;
+ unsigned param;
+ int dir;
- return p_block;
-}
+ snd_pcm_hw_params_alloca (&hw);
+ snd_pcm_hw_params_any (pcm, hw);
+ Dump (demux, "initial hardware setup:\n", snd_pcm_hw_params_dump, hw);
-/*****************************************************************************
- * OpenAudioDev: open and set up the audio device and probe for capabilities
- *****************************************************************************/
-static int OpenAudioDevAlsa( demux_t *p_demux )
-{
- demux_sys_t *p_sys = p_demux->p_sys;
- const char *psz_device = p_sys->psz_device;
- p_sys->p_alsa_pcm = NULL;
- snd_pcm_hw_params_t *p_hw_params = NULL;
- snd_pcm_uframes_t buffer_size;
- snd_pcm_uframes_t chunk_size;
-
- /* ALSA */
- int i_err;
-
- if( ( i_err = snd_pcm_open( &p_sys->p_alsa_pcm, psz_device,
- SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0)
+ val = snd_pcm_hw_params_set_rate_resample (pcm, hw, 0);
+ if (val)
{
- msg_Err( p_demux, "Cannot open ALSA audio device %s (%s)",
- psz_device, snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "cannot disable resampling: %s", snd_strerror (val));
+ goto error;
}
- if( ( i_err = snd_pcm_nonblock( p_sys->p_alsa_pcm, 1 ) ) < 0)
+ val = snd_pcm_hw_params_set_access (pcm, hw,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (val)
{
- msg_Err( p_demux, "Cannot set ALSA nonblock (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "cannot set access mode: %s", snd_strerror (val));
+ goto error;
}
- /* Begin setting hardware parameters */
-
- if( ( i_err = snd_pcm_hw_params_malloc( &p_hw_params ) ) < 0 )
- {
- msg_Err( p_demux,
- "ALSA: cannot allocate hardware parameter structure (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
- }
+ snd_pcm_format_t format = SND_PCM_FORMAT_UNKNOWN;
+ for (size_t i = 0; i < sizeof (choices) / sizeof (choices[0]); i++)
+ if (snd_pcm_hw_params_test_format (pcm, hw, choices[i]) == 0)
+ {
+ val = snd_pcm_hw_params_set_format (pcm, hw, choices[i]);
+ if (val)
+ {
+ msg_Err (demux, "cannot set sample format: %s",
+ snd_strerror (val));
+ goto error;
+ }
+ format = choices[i];
+ break;
+ }
- if( ( i_err = snd_pcm_hw_params_any( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
+ if (format == SND_PCM_FORMAT_UNKNOWN)
{
- msg_Err( p_demux,
- "ALSA: cannot initialize hardware parameter structure (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "no supported sample format");
+ goto error;
}
- /* Set Interleaved access */
- if( ( i_err = snd_pcm_hw_params_set_access( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
- {
- msg_Err( p_demux, "ALSA: cannot set access type (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
- }
+ assert ((size_t)format < (sizeof (formats) / sizeof (formats[0])));
+ es_format_Init (&fmt, AUDIO_ES, formats[format]);
+ fmt.audio.i_format = fmt.i_codec;
- /* Set 16 bit little endian */
- if( ( i_err = snd_pcm_hw_params_set_format( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_FORMAT_S16_LE ) ) < 0 )
+ param = 1 + var_InheritBool (demux, "alsa-stereo");
+ val = snd_pcm_hw_params_set_channels_max (pcm, hw, ¶m);
+ if (val)
{
- msg_Err( p_demux, "ALSA: cannot set sample format (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "cannot restrict channels count: %s",
+ snd_strerror (val));
+ goto error;
}
-
- /* Set sample rate */
-#ifdef HAVE_ALSA_NEW_API
- i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, &p_sys->i_sample_rate, NULL );
-#else
- i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, p_sys->i_sample_rate, NULL );
-#endif
- if( i_err < 0 )
+ val = snd_pcm_hw_params_set_channels_last (pcm, hw, ¶m);
+ if (val)
{
- msg_Err( p_demux, "ALSA: cannot set sample rate (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "cannot set channels count: %s", snd_strerror (val));
+ goto error;
}
-
- /* Set channels */
- unsigned int channels = p_sys->b_stereo ? 2 : 1;
- if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 )
+ assert (param > 0);
+ assert (param < (sizeof (channel_maps) / sizeof (channel_maps[0])));
+ fmt.audio.i_channels = param;
+ fmt.audio.i_original_channels =
+ fmt.audio.i_physical_channels = channel_maps[param - 1];
+
+ param = var_InheritInteger (demux, "alsa-samplerate");
+ val = snd_pcm_hw_params_set_rate_max (pcm, hw, ¶m, NULL);
+ if (val)
{
- channels = ( channels==1 ) ? 2 : 1;
- msg_Warn( p_demux, "ALSA: cannot set channel count (%s). "
- "Trying with channels=%d",
- snd_strerror( i_err ),
- channels );
- if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 )
- {
- msg_Err( p_demux, "ALSA: cannot set channel count (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
- }
- p_sys->b_stereo = ( channels == 2 );
+ msg_Err (demux, "cannot restrict rate to %u Hz or less: %s", 192000,
+ snd_strerror (val));
+ goto error;
}
-
- /* Set metrics for buffer calculations later */
- unsigned int buffer_time;
- if( ( i_err = snd_pcm_hw_params_get_buffer_time_max(p_hw_params, &buffer_time, 0) ) < 0 )
+ val = snd_pcm_hw_params_set_rate_last (pcm, hw, ¶m, &dir);
+ if (val)
{
- msg_Err( p_demux, "ALSA: cannot get buffer time max (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "cannot set sample rate: %s", snd_strerror (val));
+ goto error;
}
- if( buffer_time > 500000 ) buffer_time = 500000;
-
- /* Set period time */
- unsigned int period_time = buffer_time / 4;
-#ifdef HAVE_ALSA_NEW_API
- i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, &period_time, 0 );
-#else
- i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, period_time, 0 );
-#endif
- if( i_err < 0 )
+ if (dir)
+ msg_Warn (demux, "sample rate is not integral");
+ fmt.audio.i_rate = param;
+ sys->rate = param;
+
+ sys->start = mdate ();
+ sys->caching = INT64_C(1000) * var_InheritInteger (demux, "live-caching");
+ param = sys->caching;
+ val = snd_pcm_hw_params_set_buffer_time_near (pcm, hw, ¶m, NULL);
+ if (val)
{
- msg_Err( p_demux, "ALSA: cannot set period time (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "cannot set buffer duration: %s", snd_strerror (val));
+ goto error;
}
- /* Set buffer time */
-#ifdef HAVE_ALSA_NEW_API
- i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, &buffer_time, 0 );
-#else
- i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, buffer_time, 0 );
-#endif
- if( i_err < 0 )
+ param /= 4;
+ val = snd_pcm_hw_params_set_period_time_near (pcm, hw, ¶m, NULL);
+ if (val)
{
- msg_Err( p_demux, "ALSA: cannot set buffer time (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "cannot set period: %s", snd_strerror (val));
+ goto error;
}
- /* Apply new hardware parameters */
- if( ( i_err = snd_pcm_hw_params( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
+ val = snd_pcm_hw_params_get_period_size (hw, &sys->period_size, &dir);
+ if (val)
{
- msg_Err( p_demux, "ALSA: cannot set hw parameters (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ msg_Err (demux, "cannot get period size: %s", snd_strerror (val));
+ goto error;
}
+ if (dir > 0)
+ sys->period_size++;
- /* Get various buffer metrics */
- snd_pcm_hw_params_get_period_size( p_hw_params, &chunk_size, 0 );
- snd_pcm_hw_params_get_buffer_size( p_hw_params, &buffer_size );
- if( chunk_size == buffer_size )
+ /* Commit hardware parameters */
+ val = snd_pcm_hw_params (pcm, hw);
+ if (val)
{
- msg_Err( p_demux,
- "ALSA: period cannot equal buffer size (%lu == %lu)",
- chunk_size, buffer_size);
- goto adev_fail;
+ msg_Err (demux, "cannot commit hardware parameters: %s",
+ snd_strerror (val));
+ goto error;
}
+ Dump (demux, "final HW setup:\n", snd_pcm_hw_params_dump, hw);
- int bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
- int bits_per_frame = bits_per_sample * channels;
-
- p_sys->i_alsa_chunk_size = chunk_size;
- p_sys->i_alsa_frame_size = bits_per_frame / 8;
- p_sys->i_max_frame_size = chunk_size * bits_per_frame / 8;
-
- snd_pcm_hw_params_free( p_hw_params );
- p_hw_params = NULL;
+ /* Kick recording */
+ aout_FormatPrepare (&fmt.audio);
+ sys->es = es_out_Add (demux->out, &fmt);
+ demux->p_sys = sys;
- /* Prep device */
- if( ( i_err = snd_pcm_prepare( p_sys->p_alsa_pcm ) ) < 0 )
+ if (vlc_clone (&sys->thread, Thread, demux, VLC_THREAD_PRIORITY_INPUT))
{
- msg_Err( p_demux,
- "ALSA: cannot prepare audio interface for use (%s)",
- snd_strerror( i_err ) );
- goto adev_fail;
+ es_out_Del (demux->out, sys->es);
+ goto error;
}
- snd_pcm_start( p_sys->p_alsa_pcm );
-
+ demux->pf_demux = NULL;
+ demux->pf_control = Control;
return VLC_SUCCESS;
-
- adev_fail:
-
- if( p_hw_params ) snd_pcm_hw_params_free( p_hw_params );
- if( p_sys->p_alsa_pcm ) snd_pcm_close( p_sys->p_alsa_pcm );
- p_sys->p_alsa_pcm = NULL;
-
+error:
+ snd_pcm_close (pcm);
+ free (sys);
return VLC_EGENERIC;
-
-}
-
-static int OpenAudioDev( demux_t *p_demux )
-{
- demux_sys_t *p_sys = p_demux->p_sys;
- if( OpenAudioDevAlsa( p_demux ) != VLC_SUCCESS )
- return VLC_EGENERIC;
-
- msg_Dbg( p_demux, "opened adev=`%s' %s %dHz",
- p_sys->psz_device, p_sys->b_stereo ? "stereo" : "mono",
- p_sys->i_sample_rate );
-
- es_format_t fmt;
- es_format_Init( &fmt, AUDIO_ES, VLC_FOURCC('a','r','a','w') );
-
- fmt.audio.i_channels = p_sys->b_stereo ? 2 : 1;
- fmt.audio.i_rate = p_sys->i_sample_rate;
- fmt.audio.i_bitspersample = 16;
- fmt.audio.i_blockalign = fmt.audio.i_channels * fmt.audio.i_bitspersample / 8;
- fmt.i_bitrate = fmt.audio.i_channels * fmt.audio.i_rate * fmt.audio.i_bitspersample;
-
- msg_Dbg( p_demux, "new audio es %d channels %dHz",
- fmt.audio.i_channels, fmt.audio.i_rate );
-
- p_sys->p_es = es_out_Add( p_demux->out, &fmt );
-
- return VLC_SUCCESS;
}
-/*****************************************************************************
- * ProbeAudioDevAlsa: probe audio for capabilities
- *****************************************************************************/
-static bool ProbeAudioDevAlsa( demux_t *p_demux, const char *psz_device )
+static void Close (vlc_object_t *obj)
{
- int i_err;
- snd_pcm_t *p_alsa_pcm;
-
- if( ( i_err = snd_pcm_open( &p_alsa_pcm, psz_device, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0 )
- {
- msg_Err( p_demux, "cannot open device %s for ALSA audio (%s)", psz_device, snd_strerror( i_err ) );
- return false;
- }
+ demux_t *demux = (demux_t *)obj;
+ demux_sys_t *sys = demux->p_sys;
- snd_pcm_close( p_alsa_pcm );
+ vlc_cancel (sys->thread);
+ vlc_join (sys->thread, NULL);
- return true;
+ snd_pcm_close (sys->pcm);
+ free (sys);
}