/*****************************************************************************
* alsa.c : Alsa input module for vlc
*****************************************************************************
- * Copyright (C) 2002-2009 the VideoLAN team
+ * Copyright (C) 2002-2011 the VideoLAN team
* $Id$
*
* Authors: Benjamin Pracht <bigben at videolan dot org>
#include <vlc_access.h>
#include <vlc_demux.h>
#include <vlc_input.h>
+#include <vlc_fourcc.h>
+#include <vlc_aout.h>
#include <unistd.h>
#include <sys/ioctl.h>
#define STEREO_LONGTEXT N_( \
"Capture the audio stream in stereo." )
+#define FORMAT_TEXT N_( "Capture format (default s16l)" )
+#define FORMAT_LONGTEXT N_( \
+ "Capture format of audio stream." )
+
#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 HELP_TEXT N_( \
"Use alsa:// to open the default audio input. If multiple audio " \
"inputs are available, they will be listed in the vlc debug output. " \
#define ALSA_DEFAULT "hw"
#define CFG_PREFIX "alsa-"
+static const char *const ppsz_fourcc[] = {
+ "u8", "s8", "gsm", "u16l", "s16l", "u16b", "s16b",
+ "u24l", "s24l", "u24b", "s24b", "u32l", "s32l",
+ "u32b", "s32b", "f32l", "f32b", "f64l", "f64b"
+};
+static const char *const ppsz_fourcc_text[] = {
+ N_("PCM U8"), N_("PCM S8"), N_("GSM Audio"),
+ N_("PCM U16 LE"), N_("PCM S16 LE"),
+ N_("PCM U16 BE"), N_("PCM S16 BE"),
+ N_("PCM U24 LE"), N_("PCM S24 LE"),
+ N_("PCM U24 BE"), N_("PCM S24 BE"),
+ N_("PCM U32 LE"), N_("PCM S32 LE"),
+ N_("PCM U32 BE"), N_("PCM S32 BE"),
+ N_("PCM F32 LE"), N_("PCM F32 BE"),
+ N_("PCM F64 LE"), N_("PCM F64 BE")
+};
+
vlc_module_begin()
set_shortname( N_("ALSA") )
set_description( N_("ALSA audio capture input") )
set_capability( "access_demux", 10 )
set_callbacks( DemuxOpen, DemuxClose )
- add_bool( CFG_PREFIX "stereo", true, NULL, STEREO_TEXT, STEREO_LONGTEXT,
+ add_bool( CFG_PREFIX "stereo", true, STEREO_TEXT, STEREO_LONGTEXT,
true )
+ add_string( CFG_PREFIX "format", "s16l", FORMAT_TEXT,
+ FORMAT_LONGTEXT, true )
+ change_string_list( ppsz_fourcc, ppsz_fourcc_text, 0 )
add_integer( CFG_PREFIX "samplerate", 48000, SAMPLERATE_TEXT,
SAMPLERATE_LONGTEXT, true )
- add_integer( CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
- CACHING_TEXT, CACHING_LONGTEXT, true )
vlc_module_end()
/*****************************************************************************
struct demux_sys_t
{
/* Audio */
- int i_cache;
unsigned int i_sample_rate;
bool b_stereo;
+ vlc_fourcc_t i_format;
size_t i_max_frame_size;
block_t *p_block;
es_out_id_t *p_es;
p_sys->i_sample_rate = var_InheritInteger( p_demux, CFG_PREFIX "samplerate" );
p_sys->b_stereo = var_InheritBool( p_demux, CFG_PREFIX "stereo" );
- p_sys->i_cache = var_InheritInteger( p_demux, CFG_PREFIX "caching" );
p_sys->p_es = NULL;
p_sys->p_block = NULL;
p_sys->i_next_demux_date = -1;
+ char *psz_format = var_InheritString( p_demux, CFG_PREFIX "format" );
+ p_sys->i_format = vlc_fourcc_GetCodecFromString( AUDIO_ES, psz_format );
+ free( psz_format );
+
const char *psz_device = NULL;
if( p_demux->psz_location && *p_demux->psz_location )
psz_device = p_demux->psz_location;
return VLC_SUCCESS;
case DEMUX_GET_PTS_DELAY:
- *va_arg( args, int64_t * ) = (int64_t)p_sys->i_cache * 1000;
+ *va_arg( args, int64_t * ) =
+ INT64_C(1000) * var_InheritInteger( p_demux, "live-caching" );
return VLC_SUCCESS;
case DEMUX_GET_TIME:
do
{
+ 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, p_block );
p_block = NULL;
}
- /* Wait for data */
- int i_wait = snd_pcm_wait( p_sys->p_alsa_pcm, 10 ); /* See poll() comment in oss.c */
- 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 );
return 1;
}
-
/*****************************************************************************
* GrabAudio: Grab an audio frame
*****************************************************************************/
if( !p_block )
{
msg_Warn( p_demux, "cannot get block" );
- return 0;
+ return NULL;
}
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 );
+ i_read = snd_pcm_readi( p_sys->p_alsa_pcm, p_block->p_buffer,
+ p_sys->i_alsa_chunk_size );
+ if( i_read == -EAGAIN )
+ {
+ snd_pcm_wait( p_sys->p_alsa_pcm, 10 ); /* See poll() comment in oss.c */
+ return NULL;
+ }
+
+ if( i_read < 0 )
+ i_read = snd_pcm_recover( p_sys->p_alsa_pcm, i_read, 0 );
+
if( i_read <= 0 )
{
- int i_resume;
switch( i_read )
{
+ case 0: /* state recovered or no data */
+ return NULL;
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;
+ snd_pcm_wait( p_sys->p_alsa_pcm, 10 ); /* See poll() comment in oss.c */
+ return NULL;
default:
- msg_Err( p_demux, "Failed to read alsa frame (%s)", snd_strerror( i_read ) );
- return 0;
+ msg_Err( p_demux, "Failed to read alsa frame (%s)",
+ snd_strerror( i_read ) );
+ return NULL;
}
}
- else
- {
- /* convert from frames to bytes */
- i_read *= p_sys->i_alsa_frame_size;
- }
- if( i_read <= 0 ) return 0;
+ /* convert from frames to bytes */
+ i_read *= p_sys->i_alsa_frame_size;
p_block->i_buffer = i_read;
p_sys->p_block = 0;
return p_block;
}
+static snd_pcm_format_t GetAlsaPCMFormat( demux_t *p_demux, const vlc_fourcc_t i_format )
+{
+ demux_sys_t *p_sys = p_demux->p_sys;
+
+ switch( i_format )
+ {
+ case VLC_CODEC_U8: return SND_PCM_FORMAT_U8;
+ case VLC_CODEC_S8: return SND_PCM_FORMAT_S8;
+
+ case VLC_CODEC_GSM: return SND_PCM_FORMAT_GSM;
+
+ case VLC_CODEC_U16L: return SND_PCM_FORMAT_U16_LE;
+ case VLC_CODEC_S16L: return SND_PCM_FORMAT_S16_LE;
+ case VLC_CODEC_U16B: return SND_PCM_FORMAT_U16_BE;
+ case VLC_CODEC_S16B: return SND_PCM_FORMAT_S16_BE;
+
+ case VLC_CODEC_U24L: return SND_PCM_FORMAT_U24_3LE;
+ case VLC_CODEC_S24L: return SND_PCM_FORMAT_S24_3LE;
+ case VLC_CODEC_U24B: return SND_PCM_FORMAT_U24_3BE;
+ case VLC_CODEC_S24B: return SND_PCM_FORMAT_S24_3BE;
+
+ case VLC_CODEC_U32L: return SND_PCM_FORMAT_U32_LE;
+ case VLC_CODEC_U32B: return SND_PCM_FORMAT_U32_BE;
+ case VLC_CODEC_S32L: return SND_PCM_FORMAT_S32_LE;
+ case VLC_CODEC_S32B: return SND_PCM_FORMAT_S32_BE;
+ case VLC_CODEC_F32L: return SND_PCM_FORMAT_FLOAT_LE;
+ case VLC_CODEC_F32B: return SND_PCM_FORMAT_FLOAT_BE;
+
+ case VLC_CODEC_F64L: return SND_PCM_FORMAT_FLOAT64_LE;
+ case VLC_CODEC_F64B: return SND_PCM_FORMAT_FLOAT64_BE;
+
+ default:
+ msg_Err( p_demux, "ALSA: unsupported sample format '%s' falling back to 's16l'",
+ (const char *)&i_format );
+ p_sys->i_format = VLC_CODEC_S16L;
+ }
+
+ return SND_PCM_FORMAT_S16_LE;
+}
+
/*****************************************************************************
* OpenAudioDev: open and set up the audio device and probe for capabilities
*****************************************************************************/
demux_sys_t *p_sys = p_demux->p_sys;
p_sys->p_alsa_pcm = NULL;
snd_pcm_hw_params_t *p_hw_params = NULL;
+ snd_pcm_format_t i_alsa_pcm_format;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t chunk_size;
}
/* 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 )
+ 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;
}
- /* 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 )
+ /* Set capture format, default is signed 16 bit little endian */
+ i_alsa_pcm_format = GetAlsaPCMFormat( p_demux, p_sys->i_format );
+ if( ( i_err = snd_pcm_hw_params_set_format( p_sys->p_alsa_pcm, p_hw_params,
+ i_alsa_pcm_format ) ) < 0 )
{
msg_Err( p_demux, "ALSA: cannot set sample format (%s)",
snd_strerror( i_err ) );
}
/* Set sample rate */
- i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, &p_sys->i_sample_rate, NULL );
+ i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params,
+ &p_sys->i_sample_rate, NULL );
if( i_err < 0 )
{
msg_Err( p_demux, "ALSA: cannot set sample rate (%s)",
/* 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 )
+ 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_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 )
+ 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 ) );
/* Set period time */
unsigned int period_time = buffer_time / 4;
- i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, &period_time, 0 );
+ i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params,
+ &period_time, 0 );
if( i_err < 0 )
{
msg_Err( p_demux, "ALSA: cannot set period time (%s)",
}
/* Set buffer time */
- i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, &buffer_time, 0 );
+ i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params,
+ &buffer_time, 0 );
if( i_err < 0 )
{
msg_Err( p_demux, "ALSA: cannot set buffer time (%s)",
goto adev_fail;
}
- int bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
+ int bits_per_sample = snd_pcm_format_physical_width(i_alsa_pcm_format);
int bits_per_frame = bits_per_sample * channels;
p_sys->i_alsa_chunk_size = chunk_size;
p_sys->p_alsa_pcm = NULL;
return VLC_EGENERIC;
-
}
static int OpenAudioDev( demux_t *p_demux, const char *psz_device )
if( OpenAudioDevAlsa( p_demux, psz_device ) != VLC_SUCCESS )
return VLC_EGENERIC;
- msg_Dbg( p_demux, "opened adev=`%s' %s %dHz",
+ msg_Dbg( p_demux, "opened adev=`%s' %s %dHz codec '%s'",
psz_device, p_sys->b_stereo ? "stereo" : "mono",
- p_sys->i_sample_rate );
+ p_sys->i_sample_rate,
+ vlc_fourcc_GetDescription( AUDIO_ES, p_sys->i_format ) );
es_format_t fmt;
- es_format_Init( &fmt, AUDIO_ES, VLC_FOURCC('a','r','a','w') );
+ es_format_Init( &fmt, AUDIO_ES, p_sys->i_format );
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_bitspersample = aout_BitsPerSample( p_sys->i_format );
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;
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