Updated: 2022/Sep/29

Please read Privacy Policy. It's for your privacy.

AUDIO(4)                     Device Drivers 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 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

     /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.  It 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 sets the track to manipulate
     monaural 8-bit mu-law 8000Hz.  When /dev/sound is opened, it maintains
     the audio format and pause/unpause state of the most recently opened
     track.  In all other respects /dev/audio and /dev/sound are identical.

     On a full-duplex device, reads and writes may operate concurrently
     without interference.

     On a half-duplex device, if there are any recording descriptors already,
     opening with write mode will fail.  Similarly, if there are any playback
     descriptors already, opening with read mode will fail.  If both playback
     and recording are requested on a half-duplex device, it will be treated
     as playback mode.

     On either type of device, opening with write mode will start in playback
     mode, opening with read mode will start in recording mode.

     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 a
     reading process does not call read(2) frequently enough, it will simply
     miss samples.

     The audio driver supports track multiplexing.  All sampling devices can
     be opened at any time without interference.  For playback, all tracks
     opened simultaneously are mixed, even if their specified format is
     different.  For recording, recorded data is distributed to all opened
     tracks, even if their specified format is different.  To achieve this,
     the audio driver has a small efficient encoding converter, a channel
     mixer, and a frequency converter.  The frequency conversion adapts the
     simplest way (interpolation method for upward, and simple thinning method
     for downward) due to restriction in kernel resources and processing time.
     It will work well in most case but don't expect excessive quality.

     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).  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 mmap'ped buffer appears
     as a block of memory of size buffersize (as available via AUDIO_GETINFO
     or AUDIO_GETBUFINFO).  The audio driver will continuously move data from
     this buffer from/to the mixing buffer, 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.  Note
     that mmap(2) no longer maps hardware buffers directly.  Now it is
     achieved by emulation so don't expect any improvements excessively rather
     than normal write(2).  For historical reasons, only encodings that are
     not set AUDIO_ENCODINGFLAG_EMULATED are able to mmap(2).

     The audio device, like most devices, can be used in select(2), 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 on this track, and restarts
             recording and playback as appropriate for the current sampling

     AUDIO_PERROR (int)

     AUDIO_RERROR (int)
             This command fetches the count of dropped output (input) bytes
             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 bytes 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.

     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_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.
             The encoding returned by the command is user accessible encoding
             and is not hardware supported encoding.

             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_GETFD (int)
             This command is obsolete.

     AUDIO_SETFD (int)
             This command is obsolete.

             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

             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 parameters
             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 track mixer 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_t 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.  For historical reasons, the audio_info structure has
             three different layer's parameters: track, track mixer and
             hardware rich mixer.

             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 [HWmixer] */
                     /* BSD extensions */
                     u_int   blocksize;      /* read/write block size [track] */
                     u_int   hiwat;          /* output high water mark [track] */
                     u_int   lowat;          /* output low water mark [track] */
                     u_int   _ispare1;
                     u_int   mode;           /* current operation mode [track] */
             #define AUMODE_PLAY     0x01
             #define AUMODE_RECORD   0x02
             #define AUMODE_PLAY_ALL 0x04    /* Not used anymore */
             } 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 indicates current operation mode, either one of
             AUMODE_PLAY or AUMODE_RECORD.  These two flags can not be changed
             once this descriptor is opened.  For playback mode, the obsolete
             AUMODE_PLAY_ALL flag can be set but has no effect.

             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.  Normally the blocksize is calculated to correspond to
             the value of the hw.audioX.blk_ms sysctl and is recalculated when
             the encoding parameters change.  If the descriptor is opened for
             read only, blocksize indicates the blocksize for the recording
             track.  Otherwise, blocksize indicates the blocksize for the
             playback track.

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

             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:

             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 audio driver accepts the following formats.  encoding and
             precision are one of the values obtained by AUDIO_GETENC,
             regardless of formats supported by underlying driver.  frequency
             ranges from 1000Hz to 192000Hz, regardless of frequency (ranges)
             supported by underlying driver.  channels depends your underlying
             driver.  If the underlying driver only supports monaural
             (1channel) or stereo (2channels), you can specify 1 or 2
             regardless of number of channels supported by underlying driver.
             If the underlying driver supports three or more channels, you can
             specify the number of channels supported by the underlying driver
             or less.

             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 AUDIO_MID_BALANCE.

             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

             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 bytes 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.

     AUDIO_QUERYFORMAT (audio_format_query_t)
             This command enumerates formats supported by the hardware.
             Similarly to AUDIO_GETENC, to query all the supported formats,
             start with an index field of 0 and continue with successive
             formats (1, 2, ...) until the command returns an error.

             typedef struct audio_format_query {
                     u_int   index;
                     struct audio_format fmt;
             } audio_format_query_t;

     AUDIO_GETFORMAT (audio_info_t)
             This command fetches the current hardware format.  Only the
             following members in audio_info_t are used.  Members which are
             not listed here or belong in invalid direction are filled by -1.










             mode indicates which direction is valid.

     AUDIO_SETFORMAT (audio_info_t)
             This command sets the hardware format.  It will fail if there are
             any opened descriptors.  So obviously, it must be issued on
             /dev/audioctl.  Similarly to AUDIO_GETFORMAT, only above members
             in audio_info_t are used.  Members which is not listed or belong
             in invalid direction are ignored.  The parameters can be chosen
             from the choices obtained by AUDIO_QUERYFORMAT.

     AUDIO_GETCHAN (int)
             This command is obsolete.

     AUDIO_SETCHAN (int)
             This command is obsolete.

     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)

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

     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 argument.

             #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
             #define AUDIO_MIXER_LEVEL_RIGHT 1

             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 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.
             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
             recording, 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 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
             control types, the channel count is returned; the units name
             specifies what the level controls (typical values are
             AudioNvolume, AudioNtreble, AudioNbass).

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


     audiocfg(1), audioctl(1), audioplay(1), audiorecord(1), mixerctl(1),
     ioctl(2), ossaudio(3), acorn32/vidcaudio(4), arcofi(4), aria(4),
     auacer(4), audiocs(4), auich(4), auixp(4), autri(4), auvia(4), bba(4),
     btsco(4), clcs(4), clct(4), cmpci(4), dreamcast/aica(4), eap(4),
     emuxki(4), esa(4), esm(4), eso(4), ess(4), fms(4), gcscaudio(4), gus(4),
     guspnp(4), hdafg(4), hdaudio(4), hppa/harmony(4), macppc/awacs(4),
     macppc/snapper(4), midi(4), neo(4), pad(4), pas(4), radio(4), sb(4),
     sgimips/haltwo(4), sgimips/mavb(4), sparc/audioamd(4), sparc/dbri(4),
     sv(4), uaudio(4), wss(4), x68k/vs(4), yds(4), ym(4)

     Support for virtual channels and mixing first appeared in NetBSD 8.0.

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

NetBSD 10.99                    March 28, 2020                    NetBSD 10.99