--- /dev/null
+/*****************************************************************************
+ * alsa.c : Alsa input module for vlc
+ *****************************************************************************
+ * Copyright (C) 2002-2009 the VideoLAN team
+ * $Id$
+ *
+ * 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
+ * (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.
+ *
+ * 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
+ *****************************************************************************/
+
+#ifdef HAVE_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 <vlc_vout.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 <poll.h>
+
+/*****************************************************************************
+ * Module descriptior
+ *****************************************************************************/
+
+static int DemuxOpen ( vlc_object_t * );
+static void DemuxClose( vlc_object_t * );
+
+#define STEREO_TEXT N_( "Stereo" )
+#define STEREO_LONGTEXT N_( \
+ "Capture the audio stream in stereo." )
+
+#define SAMPLERATE_TEXT N_( "Samplerate" )
+#define SAMPLERATE_LONGTEXT N_( \
+ "Samplerate of the captured audio stream, in Hz (eg: 11025, 22050, 44100, 48000)" )
+
+#define CACHING_TEXT N_("Caching value in ms")
+#define CACHING_LONGTEXT N_( \
+ "Caching value for Alsa captures. This " \
+ "value should be set in milliseconds." )
+
+#define ALSA_DEFAULT "hw"
+#define CFG_PREFIX "alsa-"
+
+vlc_module_begin();
+ set_shortname( N_("Alsa") );
+ set_description( N_("Alsa audio capture input") );
+ set_category( CAT_INPUT );
+ set_subcategory( SUBCAT_INPUT_ACCESS );
+
+ add_shortcut( "alsa" );
+ set_capability( "access_demux", 10 );
+ set_callbacks( DemuxOpen, DemuxClose );
+
+ 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();
+
+/*****************************************************************************
+ * Access: local prototypes
+ *****************************************************************************/
+
+static int DemuxControl( demux_t *, int, va_list );
+
+static int Demux( demux_t * );
+
+static block_t* GrabAudio( demux_t *p_demux );
+
+static int OpenAudioDev( vlc_object_t *, demux_sys_t * );
+static bool ProbeAudioDevAlsa( vlc_object_t *, const char *psz_device );
+
+struct demux_sys_t
+{
+ const char *psz_device; /* Alsa device from MRL */
+ int i_fd_audio;
+
+ /* Audio */
+ int i_pts;
+ unsigned int i_sample_rate;
+ bool b_stereo;
+ size_t i_audio_max_frame_size;
+ block_t *p_block_audio;
+ es_out_id_t *p_es_audio;
+
+ int i_audio_method;
+
+ /* ALSA Audio */
+ snd_pcm_t *p_alsa_pcm;
+ size_t i_alsa_frame_size;
+ int i_alsa_chunk_size;
+};
+
+static int FindMainDevice( vlc_object_t *p_this, demux_sys_t *p_sys )
+{
+ msg_Dbg( p_this, "opening device '%s'", p_sys->psz_device );
+ if( ProbeAudioDevAlsa( p_this, p_sys->psz_device ) )
+ {
+ msg_Dbg( p_this, "'%s' is an audio device", p_sys->psz_device );
+ p_sys->i_fd_audio = OpenAudioDev( p_this, p_sys );
+ }
+
+ if( p_sys->i_fd_audio < 0 )
+ return VLC_EGENERIC;
+ return VLC_SUCCESS;
+}
+
+/*****************************************************************************
+ * 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_pts = var_CreateGetInteger( p_demux, CFG_PREFIX "caching" );
+ p_sys->psz_device = NULL;
+ p_sys->i_fd_audio = -1;
+ p_sys->p_es_audio = NULL;
+ p_sys->p_block_audio = NULL;
+
+ if( p_demux->psz_path && *p_demux->psz_path )
+ p_sys->psz_device = p_demux->psz_path;
+ else
+ p_sys->psz_device = ALSA_DEFAULT;
+ msg_Err( p_this, "Device is %s", p_sys->psz_device );
+
+ if( FindMainDevice( p_this, p_sys ) != VLC_SUCCESS )
+ {
+ DemuxClose( p_this );
+ return VLC_EGENERIC;
+ }
+
+ return VLC_SUCCESS;
+}
+
+/*****************************************************************************
+ * Close: close device, free resources
+ *****************************************************************************/
+static void DemuxClose( vlc_object_t *p_this )
+{
+ demux_t *p_demux = (demux_t *)p_this;
+ demux_sys_t *p_sys = p_demux->p_sys;
+
+ if( p_sys->p_alsa_pcm )
+ {
+ snd_pcm_close( p_sys->p_alsa_pcm );
+ p_sys->i_fd_audio = -1;
+ }
+ if( p_sys->i_fd_audio >= 0 ) close( p_sys->i_fd_audio );
+
+ if( p_sys->p_block_audio ) block_Release( p_sys->p_block_audio );
+ free( p_sys );
+}
+
+/*****************************************************************************
+ * 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;
+
+ switch( i_query )
+ {
+ /* Special for access_demux */
+ 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_pts * 1000;
+ return VLC_SUCCESS;
+
+ case DEMUX_GET_TIME:
+ pi64 = (int64_t*)va_arg( args, int64_t * );
+ *pi64 = mdate();
+ return VLC_SUCCESS;
+
+ /* TODO implement others */
+ default:
+ return VLC_EGENERIC;
+ }
+
+ return VLC_EGENERIC;
+}
+
+/*****************************************************************************
+ * Demux: Processes the audio frame
+ *****************************************************************************/
+static int Demux( demux_t *p_demux )
+{
+ demux_sys_t *p_sys = p_demux->p_sys;
+
+ struct pollfd fd;
+ fd.fd = p_sys->i_fd_audio;
+ fd.events = POLLIN|POLLPRI;
+ fd.revents = 0;
+
+ /* Wait for data */
+ if( poll( &fd, 1, 500 ) ) /* Timeout after 0.5 seconds since I don't know if pf_demux can be blocking. */
+ {
+ if( fd.revents & (POLLIN|POLLPRI) )
+ {
+ block_t *p_block = GrabAudio( p_demux );
+ if( p_block )
+ {
+ es_out_Control( p_demux->out, ES_OUT_SET_PCR, p_block->i_pts );
+ es_out_Send( p_demux->out, p_sys->p_es_audio, p_block );
+ }
+ }
+ }
+
+ return 1;
+}
+
+
+/*****************************************************************************
+ * GrabAudio: Grab an audio frame
+ *****************************************************************************/
+static block_t* GrabAudio( demux_t *p_demux )
+{
+ demux_sys_t *p_sys = p_demux->p_sys;
+ int i_read = 0, i_correct;
+ block_t *p_block;
+
+ printf("%s %d\n",__func__,__LINE__);
+ if( p_sys->p_block_audio ) p_block = p_sys->p_block_audio;
+ else p_block = block_New( p_demux, p_sys->i_audio_max_frame_size );
+
+ if( !p_block )
+ {
+ msg_Warn( p_demux, "cannot get block" );
+ return 0;
+ }
+
+ p_sys->p_block_audio = 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_audio = 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 )
+ {
+ size_t i_correction_delta = delay * p_sys->i_alsa_frame_size;
+ /* Test for overrun */
+ if( i_correction_delta > p_sys->i_audio_max_frame_size )
+ {
+ msg_Warn( p_demux, "ALSA read overrun (%zu > %zu)",
+ i_correction_delta, p_sys->i_audio_max_frame_size );
+ i_correction_delta = p_sys->i_audio_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 );
+ }
+
+ /* 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;
+
+ return p_block;
+}
+
+/*****************************************************************************
+ * OpenAudioDev: open and set up the audio device and probe for capabilities
+ *****************************************************************************/
+static int OpenAudioDevAlsa( vlc_object_t *p_this, demux_sys_t *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)
+ {
+ msg_Err( p_this, "Cannot open ALSA audio device %s (%s)",
+ psz_device, snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ if( ( i_err = snd_pcm_nonblock( p_sys->p_alsa_pcm, 1 ) ) < 0)
+ {
+ msg_Err( p_this, "Cannot set ALSA nonblock (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* Begin setting hardware parameters */
+
+ if( ( i_err = snd_pcm_hw_params_malloc( &p_hw_params ) ) < 0 )
+ {
+ msg_Err( p_this,
+ "ALSA: cannot allocate hardware parameter structure (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ if( ( i_err = snd_pcm_hw_params_any( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
+ {
+ msg_Err( p_this,
+ "ALSA: cannot initialize hardware parameter structure (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* 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_this, "ALSA: cannot set access type (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* 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 )
+ {
+ msg_Err( p_this, "ALSA: cannot set sample format (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* 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 )
+ {
+ msg_Err( p_this, "ALSA: cannot set sample rate (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* 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 )
+ {
+ channels = ( channels==1 ) ? 2 : 1;
+ msg_Warn( p_this, "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_this, "ALSA: cannot set channel count (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+ p_sys->b_stereo = ( channels == 2 );
+ }
+
+ /* 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 )
+ {
+ msg_Err( p_this, "ALSA: cannot get buffer time max (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+ 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 )
+ {
+ msg_Err( p_this, "ALSA: cannot set period time (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* 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 )
+ {
+ msg_Err( p_this, "ALSA: cannot set buffer time (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* Apply new hardware parameters */
+ if( ( i_err = snd_pcm_hw_params( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
+ {
+ msg_Err( p_this, "ALSA: cannot set hw parameters (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* 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 )
+ {
+ msg_Err( p_this,
+ "ALSA: period cannot equal buffer size (%lu == %lu)",
+ chunk_size, buffer_size);
+ goto adev_fail;
+ }
+
+ 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_audio_max_frame_size = chunk_size * bits_per_frame / 8;
+
+ snd_pcm_hw_params_free( p_hw_params );
+ p_hw_params = NULL;
+
+ /* Prep device */
+ if( ( i_err = snd_pcm_prepare( p_sys->p_alsa_pcm ) ) < 0 )
+ {
+ msg_Err( p_this,
+ "ALSA: cannot prepare audio interface for use (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ if( !p_sys->psz_device )
+ p_sys->psz_device = strdup( ALSA_DEFAULT );
+
+ /* Return a fake handle so other tests work */
+ return 1;
+
+ 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 );
+
+ return -1;
+
+}
+
+static int OpenAudioDev( vlc_object_t *p_this, demux_sys_t *p_sys )
+{
+ int i_fd = OpenAudioDevAlsa( p_this, p_sys );
+
+ if( i_fd < 0 )
+ return i_fd;
+
+ msg_Dbg( p_this, "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_this, "new audio es %d channels %dHz",
+ fmt.audio.i_channels, fmt.audio.i_rate );
+
+ demux_t *p_demux = (demux_t *)p_this;
+ p_sys->p_es_audio = es_out_Add( p_demux->out, &fmt );
+
+ return i_fd;
+}
+
+/*****************************************************************************
+ * ProbeAudioDevAlsa: probe audio for capabilities
+ *****************************************************************************/
+static bool ProbeAudioDevAlsa( vlc_object_t *p_this, const char *psz_device )
+{
+ 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_this, "cannot open device %s for ALSA audio (%s)", psz_device, snd_strerror( i_err ) );
+ return false;
+ }
+
+ snd_pcm_close( p_alsa_pcm );
+
+ return true;
+}
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