Commit b5615ec0 authored by dionoea's avatar dionoea

Split Alsa access module from v4l2.

parent 50ddca67
......@@ -4642,6 +4642,8 @@ then
AC_DEFINE(HAVE_ALSA_NEW_API, 1, Define if ALSA is at least rc4))
VLC_ADD_PLUGIN([alsa])
VLC_ADD_LIBS([alsa],[-lasound -lm -ldl])
VLC_ADD_PLUGIN([access_alsa])
VLC_ADD_LIBS([access_alsa],[-lasound -lm -ldl])
else
if test "${enable_alsa}" = "yes"; then
AC_MSG_ERROR([Could not find ALSA development headers])
......
......@@ -37,6 +37,7 @@ SOURCES_cdda = \
vcd/cdrom_internals.h \
$(NULL)
SOURCES_access_jack = jack.c
SOURCES_access_alsa = alsa.c
libvlc_LTLIBRARIES += \
libaccess_file_plugin.la \
......
/*****************************************************************************
* 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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