/*
* Sections based on the reference V4L2 capture example at
* http://v4l2spec.bytesex.org/spec/capture-example.html
+ *
+ * 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
*/
/*
* TODO: No mjpeg support yet.
* TODO: Tuner partial implementation.
- * TODO: Alsa input support?
+ * TODO: Alsa input support - experimental
*/
/*****************************************************************************
#include <sys/soundcard.h>
+#ifdef HAVE_ALSA
+# define ALSA_PCM_NEW_HW_PARAMS_API
+# define ALSA_PCM_NEW_SW_PARAMS_API
+# include <alsa/asoundlib.h>
+#endif
+
/*****************************************************************************
* Module descriptior
*****************************************************************************/
#define FPS_TEXT N_( "Framerate" )
#define FPS_LONGTEXT N_( "Framerate to capture, if applicable " \
"(-1 for autodetect)." )
+#define ALSA_TEXT N_( "Use Alsa" )
+#define ALSA_LONGTEXT N_( \
+ "Use ALSA instead of OSS for audio" )
#define STEREO_TEXT N_( "Stereo" )
#define STEREO_LONGTEXT N_( \
"Capture the audio stream in stereo." )
add_integer( "v4l2-hue", -1, NULL, HUE_TEXT,
HUE_LONGTEXT, VLC_TRUE );
add_float( "v4l2-fps", 0, NULL, FPS_TEXT, FPS_LONGTEXT, VLC_TRUE );
+ add_bool( "v4l2-alsa", VLC_FALSE, NULL, ALSA_TEXT, ALSA_LONGTEXT,
+ VLC_TRUE );
add_bool( "v4l2-stereo", VLC_TRUE, NULL, STEREO_TEXT, STEREO_LONGTEXT,
VLC_TRUE );
add_integer( "v4l2-samplerate", 48000, NULL, SAMPLERATE_TEXT,
vlc_bool_t IsPixelFormatSupported( demux_t *p_demux, unsigned int i_pixelformat );
+char* ResolveALSADeviceName( char *psz_device );
static int OpenVideoDev( demux_t *, char *psz_device );
static int OpenAudioDev( demux_t *, char *psz_device );
static vlc_bool_t ProbeVideoDev( demux_t *, char *psz_device );
es_out_id_t *p_es_video;
/* Audio */
- int i_sample_rate;
+ unsigned int i_sample_rate;
vlc_bool_t b_stereo;
int i_audio_max_frame_size;
block_t *p_block_audio;
es_out_id_t *p_es_audio;
+
+ /* ALSA Audio */
+ vlc_bool_t b_use_alsa;
+#ifdef HAVE_ALSA
+ snd_pcm_t *p_alsa_pcm;
+ int i_alsa_frame_size;
+ int i_alsa_chunk_size;
+#endif
};
/*****************************************************************************
p_sys->psz_requested_chroma = var_CreateGetString( p_demux, "v4l2-chroma" );
+ var_Create( p_demux, "v4l2-alsa", VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
+ var_Get( p_demux, "v4l2-alsa", &val );
+ p_sys->b_use_alsa = val.b_bool;
+
var_Create( p_demux, "v4l2-stereo", VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
var_Get( p_demux, "v4l2-stereo", &val );
p_sys->b_stereo = val.b_bool;
ParseMRL( p_demux );
+ /* Alsa support available? */
+#ifdef HAVE_ALSA
+ msg_Dbg( p_demux, "ALSA input support available" );
+#endif
+
/* Find main device (video or audio) */
if( p_sys->psz_device && *p_sys->psz_device )
{
strtol( psz_parser + strlen( "samplerate=" ),
&psz_parser, 0 );
}
+ else if( !strncmp( psz_parser, "alsa", strlen( "alsa" ) ) )
+ {
+ psz_parser += strlen( "alsa" );
+ p_sys->b_use_alsa = VLC_TRUE;
+ }
else if( !strncmp( psz_parser, "stereo", strlen( "stereo" ) ) )
{
psz_parser += strlen( "stereo" );
/* Close */
if( p_sys->i_fd_video >= 0 ) close( p_sys->i_fd_video );
- if( p_sys->i_fd_audio >= 0 ) close( p_sys->i_fd_audio );
+ if( p_sys->b_use_alsa )
+ {
+#ifdef HAVE_ALSA
+ if( p_sys->p_alsa_pcm ) snd_pcm_close( p_sys->p_alsa_pcm );
+#endif
+ }
+ else
+ {
+ 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 );
if( p_sys->psz_device ) free( p_sys->psz_device );
int i_read, i_correct;
block_t *p_block;
- /* Copied from v4l.c */
-
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 );
p_sys->p_block_audio = p_block;
- i_read = read( p_sys->i_fd_audio, p_block->p_buffer,
- p_sys->i_audio_max_frame_size );
+ if( p_sys->b_use_alsa )
+ {
+ /* ALSA */
+#ifdef HAVE_ALSA
+ i_read = snd_pcm_readi( p_sys->p_alsa_pcm, p_block->p_buffer, p_sys->i_alsa_chunk_size );
+ /* TODO: ALSA ERROR HANDLING?? xrun?? */
+#else
+ i_read = 0;
+#endif
+ }
+ else
+ {
+ /* OSS */
+ i_read = read( p_sys->i_fd_audio, p_block->p_buffer,
+ p_sys->i_audio_max_frame_size );
+ }
if( i_read <= 0 ) return 0;
/* Correct the date because of kernel buffering */
i_correct = i_read;
- if( ioctl( p_sys->i_fd_audio, SNDCTL_DSP_GETISPACE, &buf_info ) == 0 )
+ if( !p_sys->b_use_alsa )
{
- i_correct += buf_info.bytes;
+ if( ioctl( p_sys->i_fd_audio, SNDCTL_DSP_GETISPACE, &buf_info ) == 0 )
+ {
+ i_correct += buf_info.bytes;
+ }
+ }
+ else
+ {
+#ifdef HAVE_ALSA
+ /* TODO: ALSA timing */
+ /* Very experimental code... */
+ int i_err;
+ snd_pcm_sframes_t delay = 0;
+ if( ( i_err = snd_pcm_delay( p_sys->p_alsa_pcm, &delay ) ) >= 0 )
+ {
+ int 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" );
+ 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 );
+ }
+#endif
}
/* Timestamp */
}
+/*****************************************************************************
+ * ResolveALSADeviceName: Change any . to : in the ALSA device name
+ *****************************************************************************/
+char* ResolveALSADeviceName( char *psz_device )
+{
+ char* psz_alsa_name = strdup( psz_device );
+ for( unsigned int i = 0; i < strlen( psz_device ); i++ )
+ {
+ if( psz_alsa_name[i] == '.' ) psz_alsa_name[i] = ':';
+ }
+ return psz_alsa_name;
+}
+
/*****************************************************************************
* OpenAudioDev: open and set up the audio device and probe for capabilities
*****************************************************************************/
int OpenAudioDev( demux_t *p_demux, char *psz_device )
{
demux_sys_t *p_sys = p_demux->p_sys;
- int i_fd, i_format;
+ int i_fd = 0;
+ int i_format;
+#ifdef HAVE_ALSA
+ p_sys->p_alsa_pcm = NULL;
+ char* psz_alsa_device_name = ResolveALSADeviceName( psz_device );
+ snd_pcm_hw_params_t *p_hw_params = NULL;
+ snd_pcm_uframes_t buffer_size;
+ snd_pcm_uframes_t chunk_size;
+#endif
- if( (i_fd = open( psz_device, O_RDONLY | O_NONBLOCK )) < 0 )
+ if( p_sys->b_use_alsa )
{
- msg_Err( p_demux, "cannot open audio device (%m)" );
- goto adev_fail;
- }
+ /* ALSA */
- i_format = AFMT_S16_LE;
- if( ioctl( i_fd, SNDCTL_DSP_SETFMT, &i_format ) < 0
- || i_format != AFMT_S16_LE )
- {
- msg_Err( p_demux, "cannot set audio format (16b little endian) "
- "(%m)" );
- goto adev_fail;
- }
+#ifdef HAVE_ALSA
+ int i_err;
- if( ioctl( i_fd, SNDCTL_DSP_STEREO,
- &p_sys->b_stereo ) < 0 )
- {
- msg_Err( p_demux, "cannot set audio channels count (%m)" );
- goto adev_fail;
- }
+ if( ( i_err = snd_pcm_open( &p_sys->p_alsa_pcm, psz_alsa_device_name,
+ SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0)
+ {
+ msg_Err( p_demux, "Cannot open ALSA audio device %s (%s)",
+ psz_alsa_device_name,
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
- if( ioctl( i_fd, SNDCTL_DSP_SPEED,
- &p_sys->i_sample_rate ) < 0 )
+ if( ( i_err = snd_pcm_nonblock( p_sys->p_alsa_pcm, 1 ) ) < 0)
+ {
+ msg_Err( p_demux, "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_demux, "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_demux, "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_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 )
+ {
+ msg_Err( p_demux, "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_demux, "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_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 );
+ }
+
+ /* 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_demux, "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_demux, "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_demux, "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_demux, "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_demux, "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_sample / 8) * channels;
+ 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_demux, "ALSA: cannot prepare audio interface for use (%s)",
+ snd_strerror( i_err ) );
+ goto adev_fail;
+ }
+
+ /* Return a fake handle so other tests work */
+ i_fd = 1;
+
+#endif
+ }
+ else
{
- msg_Err( p_demux, "cannot set audio sample rate (%m)" );
- goto adev_fail;
+ /* OSS */
+
+ if( (i_fd = open( psz_device, O_RDONLY | O_NONBLOCK )) < 0 )
+ {
+ msg_Err( p_demux, "cannot open OSS audio device (%m)" );
+ goto adev_fail;
+ }
+
+ i_format = AFMT_S16_LE;
+ if( ioctl( i_fd, SNDCTL_DSP_SETFMT, &i_format ) < 0
+ || i_format != AFMT_S16_LE )
+ {
+ msg_Err( p_demux, "cannot set audio format (16b little endian) "
+ "(%m)" );
+ goto adev_fail;
+ }
+
+ if( ioctl( i_fd, SNDCTL_DSP_STEREO,
+ &p_sys->b_stereo ) < 0 )
+ {
+ msg_Err( p_demux, "cannot set audio channels count (%m)" );
+ goto adev_fail;
+ }
+
+ if( ioctl( i_fd, SNDCTL_DSP_SPEED,
+ &p_sys->i_sample_rate ) < 0 )
+ {
+ msg_Err( p_demux, "cannot set audio sample rate (%m)" );
+ goto adev_fail;
+ }
+
+ p_sys->i_audio_max_frame_size = 6 * 1024;
}
msg_Dbg( p_demux, "opened adev=`%s' %s %dHz",
psz_device, p_sys->b_stereo ? "stereo" : "mono",
p_sys->i_sample_rate );
- p_sys->i_audio_max_frame_size = 6 * 1024;
-
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; /* FIXME ? */
+ 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;
p_sys->p_es_audio = es_out_Add( p_demux->out, &fmt );
+#ifdef HAVE_ALSA
+ free( psz_alsa_device_name );
+#endif
+
return i_fd;
adev_fail:
if( i_fd >= 0 ) close( i_fd );
+
+#ifdef HAVE_ALSA
+ 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 );
+ free( psz_alsa_device_name );
+#endif
+
return -1;
}
{
int i_fd = 0;
int i_caps;
+ demux_sys_t *p_sys = p_demux->p_sys;
- if( ( i_fd = open( psz_device, O_RDONLY | O_NONBLOCK ) ) < 0 )
+ if( p_sys->b_use_alsa )
{
- msg_Err( p_demux, "cannot open audio device (%m)" );
- goto open_failed;
- }
+ /* ALSA */
- /* this will fail if the device is video */
- if( ioctl( i_fd, SNDCTL_DSP_GETCAPS, &i_caps ) < 0 )
- {
- msg_Err( p_demux, "cannot get audio caps (%m)" );
+#ifdef HAVE_ALSA
+ int i_err;
+ snd_pcm_t *p_alsa_pcm;
+ char* psz_alsa_device_name = ResolveALSADeviceName( psz_device );
+
+ if( ( i_err = snd_pcm_open( &p_alsa_pcm, psz_alsa_device_name, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0 )
+ {
+ msg_Err( p_demux, "cannot open device %s for ALSA audio (%s)", psz_alsa_device_name, snd_strerror( i_err ) );
+ free( psz_alsa_device_name );
+ goto open_failed;
+ }
+
+ snd_pcm_close( p_alsa_pcm );
+ free( psz_alsa_device_name );
+#else
+ msg_Err( p_demux, "ALSA support not available" );
goto open_failed;
+#endif
+ }
+ else
+ {
+ /* OSS */
+
+ if( ( i_fd = open( psz_device, O_RDONLY | O_NONBLOCK ) ) < 0 )
+ {
+ msg_Err( p_demux, "cannot open device %s for OSS audio (%m)", psz_device );
+ goto open_failed;
+ }
+
+ /* this will fail if the device is video */
+ if( ioctl( i_fd, SNDCTL_DSP_GETCAPS, &i_caps ) < 0 )
+ {
+ msg_Err( p_demux, "cannot get audio caps (%m)" );
+ goto open_failed;
+ }
+
+ if( i_fd >= 0 ) close( i_fd );
}
- if( i_fd >= 0 ) close( i_fd );
return VLC_TRUE;
open_failed:
projects / vlc / commitdiff