AUDIO(4)                  NetBSD Programmer's Manual                  AUDIO(4)

     audio - device-independent audio driver layer

     #include <sys/types.h>
     #include <sys/audioio.h>

     The audio driver provides support for various audio peripherals.  It pro-
     vides a uniform programming 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 ma-
     nipulate the audio device while it is in use.

     When /dev/audio is opened, it automatically directs the underlying driver
     to manipulate monaural 8-bit mulaw 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 playing (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 identical.

     Only one process may hold open a sampling device at a given time (al-
     though 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 discarded.  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 de-
     vice, playback mode takes precedence and recordings will get silence.  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 explicitely.  On either type of device, if the play-
     back mode is paused then silence is played instead of the provided sam-
     ples, 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 ac-
     cessed 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 ). The device driver will con-
     tinuously 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 cur-
     rently 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 device.

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

     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 play-
             back 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 (int)
             This command fetches the count of samples are queued ahead of the
             first sample in the most recent sample block written into its in-
             teger argument.
             This command suspends the calling process until all queued play-
             back 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 argument.

             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 opera-
             tion 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 hard-
             ware has a certain property the corresponding bit is set, other-

             wise it is not.  The properties can have the following values:
                     the device admits full duplex operation.
                     the device can be used with mmap(2).
                     the device can set the playing and recording encoding pa-
                     rameters independently.
     AUDIO_GETIOFFS (audio_offset_t)
     AUDIO_GETOOFFS (audio_offset_t)
             This command fetches the current offset in the input(output)
             buffer where the hardware is 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

             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_SETINFO (audio_info_t)
             Get or set audio information as encoded in the audio_info struc-

             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;
                     /* 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 */

             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
             AUDIO_GETINFO first.

             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 simultaneous record and play-

             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 ad-
             just 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 recalcu-
             lated when the encoding parameter changes, but if the blocksize
             is set explicitely this value becomes sticky, i.e., it is remains
             even when the encoding is changed.  The stickyness can be cleared
             by reopening 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_* above) */
                     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 ocurred */
                     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 settings the
             AUDIO_PROP_INDEPENDENT property is set.

             The encoding parameter can have the following values:
                     mulaw encoding, 8 bits/sample
                     alaw encoding, 8 bits/sample
                     two's complement signed linear encoding with the platform
                     byte order
                     unsigned linear encoding with the platform byte order
                     ADPCM encoding, 8 bits/sample
                     two's complement signed linear encoding with little endi-
                     an byte order
                     two's complement signed linear encoding with big endian
                     byte order
                     unsigned linear encoding with little endian byte order
                     unsigned linear encoding with little big byte order

             The gain, port and balance settings provide simple shortcuts to
             the richer mixer interface described below.  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] withe the
             normal setting at AUDIO_MID_BALANCE.
             The input port should be a combination of

                     to select microphone input.
                     to select line input.
                     to select CD input.
             The output port should be a combination of
                     to select microphone output.
                     to select headphone output.
                     to select line output.
             The available ports can be found in avail_ports.

             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 for AUDIO_GETINFO. 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 consumption/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) com-
     AUDIO_GETDEV (audio_device_t)
             This command is the same as described above for the sampling de-
     AUDIO_MIXER_READ (mixer_ctrl_t)
     AUDIO_MIXER_WRITE (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;
             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 argument.  For a mixer val-
             ue, the value field specifies both the number of channels and the
             values for each of the channels.  If the channel count does not
             match the current channel count, the attempt to change the set-
             ting may fail (depending on the hardware device driver implemen-
             tation).  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 in-
             formation into the input/output mixer_devinfo_t argument.  To
             query all the supported encodings, start with an index field of 0
             and continue with successive encodings (1, 2, ...) until the com-
             mand 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;
                             } v;
                     } un;
             } mixer_devinfo_t;
             The label field identifies the name of this particular mixer con-
             trol.  The index field may be used as the dev field in
             AUDIO_MIXER_READ and AUDIO_MIXER_WRITE commands.  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/output device selection (e.g. select recording input
             source from CD, line in, or microphone).  Set types are similar
             to enumeration types but any combination of the mask bits can be

             The mixer_class field identifies what class of control this is.
             This value is set to the index value used to query the class it-
             self.  For example, a mixer level controlling the input gain on
             the ``line in'' circuit would be a class that matches an input
             class device with the name ``Inputs'' (AudioCInputs).  Mixer con-
             trols 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.

             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 pre-
             vious value, AUDIO_MIXER_LAST is specified.

             For AUDIO_MIXER_ENUM mixer control types, the enumeration values
             and their corresponding 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 con-
             trol types, the channel count is returned; the units name speci-
             fies what the level controls (typical values are AudioNvolume,
             AudioNtreble, AudioNbass).

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


     ioctl(2),  ossaudio(3).
     For ports using the ISA bus: gus(4),  guspnp(4),  pas(4),  pss(4),
     sb(4),  wss(4).

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

NetBSD                          March 11, 1997                               7

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