AUDIO(4)                NetBSD 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 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

     /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.

     /dev/sound and /dev/audio can be opened at any time and audio sources of
     different precision and playback parameters i.e frequency will be mixed
     and played back simultaneously.

     /dev/audioctl can be used to manipulate the audio device while it is in

     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 playing (recording)
     paused.  When /dev/sound is opened, it maintains the previous audio sam-
     ple mode and record/playback mode most recently set on /dev/sound by any
     open channel.  In all other respects /dev/audio and /dev/sound are iden-

     Any process may open a sampling device at a given time.  Any number of
     devices per process and file descriptors may be shared between processes.

     Virtual channels are converted to a common format, signed linear encod-
     ing, frequency channels and precision.  These can be modified to taste by
     the following sysctl(8) variables.






     Where driverN corresponds to the underlying audio device driver and
     device number.  e.g In the case of an hdafg supported device the vari-
     ables would be: hw.hdafg0.channels, hw.hdafg0.precision, hw.hdafg0.fre-

     For best results, values close to the underlying hardware should be cho-
     sen.  These variables may only be changed when the sampling device is not
     in use.

     The hw.driverN.latency sysctl(8) variable controls the latency of the in-
     kernel mixer by varying the hardware blocksize.  It accepts a value in
     milliseconds(ms), fractional values are not allowed.  A value of zero
     will default to 150ms.

     If a static blocksize is enforced by the underlying hardware driver this
     value cannot be changed.

     For audio applications that do not specify a preferred blocksize when
     configuring the audio device, this will be the latency these applications

     For audio applications that mmap(2) the audio device for play back the
     resultant latency is a third (1/3) of the value of the hw.driverN.latency

     The hw.driverN.multiuser sysctl(8) variable determines if multiple users
     are allowed to access the sampling device.

     By default it is set to false.  This means that the sampling device may
     be only used by one user at a time.  Other users (except root) attempting
     to open the sampling device will be denied.

     If set to true, all users may access the sampling device at any time.

     Each virtual channel has a corresponding mixer:

     vchan.dacN Output volume

     vchan.micN Recording volume

     Where N is the virtual channel number.  e.g vchan.dac0 controlling play-
     back volume and vchan.mic0 controlling recording volume for the first
     virtual channel.

     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
     device, playback mode takes precedence and recordings will get silence.

     On a full-duplex device, reads and writes may operate concurrently with-
     out 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) 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:

     AUDIO_GETCHAN (int)
             This command will return the audio channel in use.

     AUDIO_SETCHAN (int)
             This command will select the audio channel for subsequent ioctl

             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 (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 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:

             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 indepen-
             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(get-
             ting) 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 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;                   /* 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 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
             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 recalcu-
             lated 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 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_* 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 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 encod-
                                        ing 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 encod-
                                        ing with little endian byte order
             AUDIO_ENCODING_SLINEAR_BE  two's complement signed linear encod-
                                        ing 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 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 samples pending;
             samples represents the total number of bytes recorded or played,
             less those that were dropped due to inadequate consumption/pro-
             duction 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

     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 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;
                                     int delta;
                             } 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.
             The (arbitrary) value set by the hardware driver may be deter-
             mined by examining the mixer_class field of the class itself, a
             mixer of type AUDIO_MIXER_CLASS.  For example, a mixer control-
             ling 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 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 devices can be distinguished from other
     mixer controls because they use a name from one of the AudioC* string


     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)



   The NetBSD audio specification

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

NetBSD 8.1                       May 15, 2018                       NetBSD 8.1

You can also request any man page by name and (optionally) by section:


Use the DEFAULT collection to view manual pages for third-party software.

©1994 Man-cgi 1.15, Panagiotis Christias
©1996-2019 Modified for NetBSD by Kimmo Suominen