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NetBSD 6.1.5 - man page for audioctl (netbsd section 4)

AUDIO(4)			   BSD Kernel Interfaces Manual 			 AUDIO(4)

     audio -- device-independent audio driver layer

     #include <sys/audioio.h>

     The audio driver provides support for various audio peripherals.  It provides a uniform pro-
     gramming interface layer above different underlying audio hardware drivers.  The audio layer
     provides full-duplex operation if the underlying hardware configuration supports it.

     There are four device files available for audio operation: /dev/audio, /dev/sound,
     /dev/audioctl, and /dev/mixer.

     /dev/audio and /dev/sound are used for recording or playback of digital samples.

     /dev/mixer is used to manipulate volume, recording source, or other audio mixer functions.

     /dev/audioctl accepts the same ioctl(2) operations as /dev/sound, but no other operations.

     In contrast to /dev/sound which has the exclusive open property /dev/audioctl can be opened
     at any time and can be used to manipulate the audio device while it is in use.

     When /dev/audio is opened, it automatically directs the underlying driver to manipulate
     monaural 8-bit mu-law samples.  In addition, if it is opened read-only (write-only) the
     device is set to half-duplex record (play) mode with recording (playing) unpaused and play-
     ing (recording) paused.  When /dev/sound is opened, it maintains the previous audio sample
     mode and record/playback mode.  In all other respects /dev/audio and /dev/sound are identi-

     Only one process may hold open a sampling device at a given time (although file descriptors
     may be shared between processes once the first open completes).

     On a half-duplex device, writes while recording is in progress will be immediately dis-
     carded.  Similarly, reads while playback is in progress will be filled with silence but
     delayed to return at the current sampling rate.  If both playback and recording are
     requested on a half-duplex device, playback mode takes precedence and recordings will get

     On a full-duplex device, reads and writes may operate concurrently without interference.  If
     a full-duplex capable audio device is opened for both reading and writing it will start in
     half-duplex play mode; full-duplex mode has to be set explicitly.

     On either type of device, if the playback mode is paused then silence is played instead of
     the provided samples, and if recording is paused then the process blocks in read(2) until
     recording is unpaused.

     If a writing process does not call write(2) frequently enough to provide samples at the pace
     the hardware consumes them silence is inserted.  If the AUMODE_PLAY_ALL mode is not set the
     writing process must provide enough data via subsequent write calls to ``catch up'' in time
     to the current audio block before any more process-provided samples will be played.  If a
     reading process does not call read(2) frequently enough, it will simply miss samples.

     The audio device is normally accessed with read(2) or write(2) calls, but it can also be
     mapped into user memory with mmap(2) (when supported by the device).  Once the device has
     been mapped it can no longer be accessed by read or write; all access is by reading and
     writing to the mapped memory.  The device appears as a block of memory of size buffersize
     (as available via AUDIO_GETINFO or AUDIO_GETBUFINFO).  The device driver will continuously
     move data from this buffer from/to the audio hardware, wrapping around at the end of the
     buffer.  To find out where the hardware is currently accessing data in the buffer the
     AUDIO_GETIOFFS and AUDIO_GETOOFFS calls can be used.  The playing and recording buffers are
     distinct and must be mapped separately if both are to be used.  Only encodings that are not
     emulated (i.e. where AUDIO_ENCODINGFLAG_EMULATED is not set) work properly for a mapped

     The audio device, like most devices, can be used in select, can be set in non-blocking mode
     and can be set (with a FIOASYNC ioctl) to send a SIGIO when I/O is possible.  The mixer
     device can be set to generate a SIGIO whenever a mixer value is changed.

     The following ioctl(2) commands are supported on the sample devices:

	     This command stops all playback and recording, clears all queued buffers, resets
	     error counters, and restarts recording and playback as appropriate for the current
	     sampling mode.

     AUDIO_RERROR (int)
	     This command fetches the count of dropped input samples into its integer argument.
	     There is no information regarding when in the sample stream they were dropped.

     AUDIO_WSEEK (u_long)
	     This command fetches the count of samples that are queued ahead of the first sample
	     in the most recent sample block written into its integer argument.

	     This command suspends the calling process until all queued playback samples have
	     been played by the hardware.

     AUDIO_GETDEV (audio_device_t)
	     This command fetches the current hardware device information into the audio_device_t

	     typedef struct audio_device {
		     char name[MAX_AUDIO_DEV_LEN];
		     char version[MAX_AUDIO_DEV_LEN];
		     char config[MAX_AUDIO_DEV_LEN];
	     } audio_device_t;

     AUDIO_GETFD (int)
	     The command returns the current setting of the full duplex mode.

     AUDIO_GETENC (audio_encoding_t)
	     This command is used iteratively to fetch sample encoding names and format_ids into
	     the input/output audio_encoding_t argument.

	     typedef struct audio_encoding {
		     int index;      /* input: nth encoding */
		     char name[MAX_AUDIO_DEV_LEN]; /* name of encoding */
		     int encoding;   /* value for encoding parameter */
		     int precision;  /* value for precision parameter */
		     int flags;
	     #define AUDIO_ENCODINGFLAG_EMULATED 1 /* software emulation mode */
	     } audio_encoding_t;

	     To query all the supported encodings, start with an index field of 0 and continue
	     with successive encodings (1, 2, ...) until the command returns an error.

     AUDIO_SETFD (int)
	     This command sets the device into full-duplex operation if its integer argument has
	     a non-zero value, or into half-duplex operation if it contains a zero value.  If the
	     device does not support full-duplex operation, attempting to set full-duplex mode
	     returns an error.

	     This command gets a bit set of hardware properties.  If the hardware has a certain
	     property the corresponding bit is set, otherwise it is not.  The properties can have
	     the following values:

	     AUDIO_PROP_FULLDUPLEX   the device admits full duplex operation.
	     AUDIO_PROP_MMAP	     the device can be used with mmap(2).
	     AUDIO_PROP_INDEPENDENT  the device can set the playing and recording encoding param-
				     eters independently.
	     AUDIO_PROP_PLAYBACK     the device is capable of audio playback.
	     AUDIO_PROP_CAPTURE      the device is capable of audio capture.

     AUDIO_GETIOFFS (audio_offset_t)

     AUDIO_GETOOFFS (audio_offset_t)
	     This command fetches the current offset in the input(output) buffer where the audio
	     hardware's DMA engine will be putting(getting) data.  It mostly useful when the
	     device buffer is available in user space via the mmap(2) call.  The information is
	     returned in the audio_offset structure.

	     typedef struct audio_offset {
		     u_int   samples;	/* Total number of bytes transferred */
		     u_int   deltablks; /* Blocks transferred since last checked */
		     u_int   offset;	/* Physical transfer offset in buffer */
	     } audio_offset_t;

     AUDIO_GETINFO (audio_info_t)

     AUDIO_GETBUFINFO (audio_info_t)

     AUDIO_SETINFO (audio_info_t)
	     Get or set audio information as encoded in the audio_info structure.

	     typedef struct audio_info {
		     struct  audio_prinfo play;   /* info for play (output) side */
		     struct  audio_prinfo record; /* info for record (input) side */
		     u_int   monitor_gain;		     /* input to output mix */
		     /* BSD extensions */
		     u_int   blocksize;      /* H/W read/write block size */
		     u_int   hiwat;	     /* output high water mark */
		     u_int   lowat;	     /* output low water mark */
		     u_int   _ispare1;
		     u_int   mode;	     /* current device mode */
	     #define AUMODE_PLAY     0x01
	     #define AUMODE_RECORD   0x02
	     #define AUMODE_PLAY_ALL 0x04    /* do not do real-time correction */
	     } audio_info_t;

	     When setting the current state with AUDIO_SETINFO, the audio_info structure should
	     first be initialized with AUDIO_INITINFO (&info) and then the particular values to
	     be changed should be set.	This allows the audio driver to only set those things
	     that you wish to change and eliminates the need to query the device with

	     The mode field should be set to AUMODE_PLAY, AUMODE_RECORD, AUMODE_PLAY_ALL, or a
	     bitwise OR combination of the three.  Only full-duplex audio devices support simul-
	     taneous record and playback.

	     hiwat and lowat are used to control write behavior.  Writes to the audio devices
	     will queue up blocks until the high-water mark is reached, at which point any more
	     write calls will block until the queue is drained to the low-water mark.  hiwat and
	     lowat set those high- and low-water marks (in audio blocks).  The default for hiwat
	     is the maximum value and for lowat 75 % of hiwat.

	     blocksize sets the current audio blocksize.  The generic audio driver layer and the
	     hardware driver have the opportunity to adjust this block size to get it within
	     implementation-required limits.  Upon return from an AUDIO_SETINFO call, the actual
	     blocksize set is returned in this field.  Normally the blocksize is calculated to
	     correspond to 50ms of sound and it is recalculated when the encoding parameter
	     changes, but if the blocksize is set explicitly this value becomes sticky, i.e., it
	     remains even when the encoding is changed.  The stickiness can be cleared by reopen-
	     ing the device or setting the blocksize to 0.

	     struct audio_prinfo {
		     u_int   sample_rate;    /* sample rate in samples/s */
		     u_int   channels;	     /* number of channels, usually 1 or 2 */
		     u_int   precision;      /* number of bits/sample */
		     u_int   encoding;	     /* data encoding (AUDIO_ENCODING_* below) */
		     u_int   gain;	     /* volume level */
		     u_int   port;	     /* selected I/O port */
		     u_long  seek;	     /* BSD extension */
		     u_int   avail_ports;    /* available I/O ports */
		     u_int   buffer_size;    /* total size audio buffer */
		     u_int   _ispare[1];
		     /* Current state of device: */
		     u_int   samples;	     /* number of samples */
		     u_int   eof;	     /* End Of File (zero-size writes) counter */
		     u_char  pause;	     /* non-zero if paused, zero to resume */
		     u_char  error;	     /* non-zero if underflow/overflow occurred */
		     u_char  waiting;	     /* non-zero if another process hangs in open */
		     u_char  balance;	     /* stereo channel balance */
		     u_char  cspare[2];
		     u_char  open;	     /* non-zero if currently open */
		     u_char  active;	     /* non-zero if I/O is currently active */

	     Note:  many hardware audio drivers require identical playback and recording sample
	     rates, sample encodings, and channel counts.  The playing information is always set
	     last and will prevail on such hardware.  If the hardware can handle different set-
	     tings the AUDIO_PROP_INDEPENDENT property is set.

	     The encoding parameter can have the following values:

	     AUDIO_ENCODING_ULAW	mu-law encoding, 8 bits/sample
	     AUDIO_ENCODING_ALAW	A-law encoding, 8 bits/sample
	     AUDIO_ENCODING_SLINEAR	two's complement signed linear encoding with the platform
					byte order
	     AUDIO_ENCODING_ULINEAR	unsigned linear encoding with the platform byte order
	     AUDIO_ENCODING_ADPCM	ADPCM encoding, 8 bits/sample
	     AUDIO_ENCODING_SLINEAR_LE	two's complement signed linear encoding with little
					endian byte order
	     AUDIO_ENCODING_SLINEAR_BE	two's complement signed linear encoding with big endian
					byte order
	     AUDIO_ENCODING_ULINEAR_LE	unsigned linear encoding with little endian byte order
	     AUDIO_ENCODING_ULINEAR_BE	unsigned linear encoding with big endian byte order
	     AUDIO_ENCODING_AC3 	Dolby Digital AC3

	     The gain, port and balance settings provide simple shortcuts to the richer mixer
	     interface described below and are not obtained by AUDIO_GETBUFINFO.  The gain should
	     be in the range [AUDIO_MIN_GAIN, AUDIO_MAX_GAIN] and the balance in the range
	     [AUDIO_LEFT_BALANCE, AUDIO_RIGHT_BALANCE] with the normal setting at

	     The input port should be a combination of:

	     AUDIO_MICROPHONE  to select microphone input.
	     AUDIO_LINE_IN     to select line input.
	     AUDIO_CD	       to select CD input.

	     The output port should be a combination of:

	     AUDIO_SPEAKER    to select speaker output.
	     AUDIO_HEADPHONE  to select headphone output.
	     AUDIO_LINE_OUT   to select line output.

	     The available ports can be found in avail_ports (AUDIO_GETBUFINFO only).

	     buffer_size is the total size of the audio buffer.  The buffer size divided by the
	     blocksize gives the maximum value for hiwat.  Currently the buffer_size can only be
	     read and not set.

	     The seek and samples fields are only used by AUDIO_GETINFO and AUDIO_GETBUFINFO.
	     seek represents the count of samples pending; samples represents the total number of
	     bytes recorded or played, less those that were dropped due to inadequate consump-
	     tion/production rates.

	     pause returns the current pause/unpause state for recording or playback.  For
	     AUDIO_SETINFO, if the pause value is specified it will either pause or unpause the
	     particular direction.

     The mixer device, /dev/mixer, may be manipulated with ioctl(2) but does not support read(2)
     or write(2).  It supports the following ioctl(2) commands:

     AUDIO_GETDEV (audio_device_t)
	     This command is the same as described above for the sampling devices.

     AUDIO_MIXER_READ (mixer_ctrl_t)

     AUDIO_MIXER_WRITE (mixer_ctrl_t)
	     These commands read the current mixer state or set new mixer state for the specified
	     device dev.  type identifies which type of value is supplied in the mixer_ctrl_t

	     #define AUDIO_MIXER_CLASS	0
	     #define AUDIO_MIXER_ENUM	1
	     #define AUDIO_MIXER_SET	2
	     #define AUDIO_MIXER_VALUE	3
	     typedef struct mixer_ctrl {
		     int dev;			     /* input: nth device */
		     int type;
		     union {
			     int ord;		     /* enum */
			     int mask;		     /* set */
			     mixer_level_t value;    /* value */
		     } un;
	     } mixer_ctrl_t;

	     #define AUDIO_MIN_GAIN  0
	     #define AUDIO_MAX_GAIN  255
	     typedef struct mixer_level {
		     int num_channels;
		     u_char level[8];		    /* [num_channels] */
	     } mixer_level_t;
	     #define AUDIO_MIXER_LEVEL_MONO  0
	     #define AUDIO_MIXER_LEVEL_LEFT  0

	     For a mixer value, the value field specifies both the number of channels and the
	     values for each channel.  If the channel count does not match the current channel
	     count, the attempt to change the setting may fail (depending on the hardware device
	     driver implementation).  For an enumeration value, the ord field should be set to
	     one of the possible values as returned by a prior AUDIO_MIXER_DEVINFO command.  The
	     type AUDIO_MIXER_CLASS is only used for classifying particular mixer device types
	     and is not used for AUDIO_MIXER_READ or AUDIO_MIXER_WRITE.

     AUDIO_MIXER_DEVINFO (mixer_devinfo_t)
	     This command is used iteratively to fetch audio mixer device information into the
	     input/output mixer_devinfo_t argument.  To query all the supported devices, start
	     with an index field of 0 and continue with successive devices (1, 2, ...) until the
	     command returns an error.

	     typedef struct mixer_devinfo {
		     int index; 	     /* input: nth mixer device */
		     audio_mixer_name_t label;
		     int type;
		     int mixer_class;
		     int next, prev;
	     #define AUDIO_MIXER_LAST	     -1
		     union {
			     struct audio_mixer_enum {
				     int num_mem;
				     struct {
					     audio_mixer_name_t label;
					     int ord;
				     } member[32];
			     } e;
			     struct audio_mixer_set {
				     int num_mem;
				     struct {
					     audio_mixer_name_t label;
					     int mask;
				     } member[32];
			     } s;
			     struct audio_mixer_value {
				     audio_mixer_name_t units;
				     int num_channels;
				     int delta;
			     } v;
		     } un;
	     } mixer_devinfo_t;

	     The label field identifies the name of this particular mixer control.  The index
	     field may be used as the dev field in AUDIO_MIXER_READ and AUDIO_MIXER_WRITE com-
	     mands.  The type field identifies the type of this mixer control.	Enumeration types
	     are typically used for on/off style controls (e.g. a mute control) or for input/out-
	     put device selection (e.g. select recording input source from CD, line in, or micro-
	     phone).  Set types are similar to enumeration types but any combination of the mask
	     bits can be used.

	     The mixer_class field identifies what class of control this is.  The (arbitrary)
	     value set by the hardware driver may be determined by examining the mixer_class
	     field of the class itself, a mixer of type AUDIO_MIXER_CLASS.  For example, a mixer
	     controlling the input gain on the line in circuit would have a mixer_class that
	     matches an input class device with the name ``inputs'' (AudioCinputs), and would
	     have a label of ``line'' (AudioNline).  Mixer controls which control audio circuitry
	     for a particular audio source (e.g. line-in, CD in, DAC output) are collected under
	     the input class, while those which control all audio sources (e.g. master volume,
	     equalization controls) are under the output class.  Hardware devices capable of
	     recording typically also have a record class, for controls that only affect record-
	     ing, and also a monitor class.

	     The next and prev may be used by the hardware device driver to provide hints for the
	     next and previous devices in a related set (for example, the line in level control
	     would have the line in mute as its ``next'' value).  If there is no relevant next or
	     previous value, AUDIO_MIXER_LAST is specified.

	     For AUDIO_MIXER_ENUM mixer control types, the enumeration values and their corre-
	     sponding names are filled in.  For example, a mute control would return appropriate
	     values paired with AudioNon and AudioNoff.  For AUDIO_MIXER_VALUE and
	     AUDIO_MIXER_SET mixer control types, the channel count is returned; the units name
	     specifies what the level controls (typical values are AudioNvolume, AudioNtreble,

     By convention, all the mixer devices can be distinguished from other mixer controls because
     they use a name from one of the AudioC* string values.


     audioctl(1), mixerctl(1), ioctl(2), ossaudio(3), midi(4), radio(4)

   ISA bus
     aria(4), ess(4), gus(4), guspnp(4), pas(4), sb(4), wss(4), ym(4)

   PCI bus
     auacer(4), auich(4), auixp(4), autri(4), auvia(4), azalia(4), clcs(4), clct(4), cmpci(4),
     eap(4), emuxki(4), esa(4), esm(4), eso(4), fms(4), neo(4), sv(4), yds(4)



     If the device is used in mmap(2) it is currently always mapped for writing (playing) due to
     VM system weirdness.

BSD					September 5, 2011				      BSD

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