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

     audio, mixer - device-independent audio driver layer

     audio* at ...

     #include <sys/types.h>
     #include <sys/ioctl.h>
     #include <sys/audioio.h>
     #include <string.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 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 configures the underlying
     driver for the hardware's default sample format, or monaural 8-bit mu-law
     if a default sample format has not been specified by the underlying
     driver.  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 format 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
     (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 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
     without interference.  If a full-duplex capable audio device is opened
     for both reading and writing, it will start in half-duplex play mode with
     recording paused.	For proper full-duplex operation, after the device is
     opened for reading and writing, full-duplex mode must be set and then
     recording must be unpaused.  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
     buffer_size (as available via AUDIO_GETINFO).  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 device.

     The audio device, like most devices, can be used in select(2), can be set
     in non-blocking mode, and can be set (with an FIOASYNC ioctl(2)) 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 *
     AUDIO_PERROR int *
	     These commands fetch the count of dropped input or output samples
	     into the int * argument, respectively.  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
	     u_long * 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 * 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 *
	     This 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 bps;	     /* value for bps parameter */
		     int msb;	     /* value for msb 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

	     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_GETIOFFS audio_offset_t *
     AUDIO_GETOOFFS audio_offset_t *
	     These commands fetch the current offset in the input (output)
	     buffer where the audio hardware's DMA engine will be putting
	     (getting) data.  They are 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_GETRRINFO audio_bufinto_t *
     AUDIO_GETPRINFO audio_bufinfo_t *
	     These commands fetch the current information about the input or
	     output buffer, respectively.  The block size, high and low water
	     marks and current position are returned in the audio_bufinfo

	     typedef struct audio_bufinfo {
		     u_int   blksize;	     /* block size */
		     u_int   hiwat;	     /* high water mark */
		     u_int   lowat;	     /* low water mark */
		     u_int   seek;	     /* current position */
	     } audio_bufinfo_t;

	     This information is mostly useful in input or output loops to
	     determine how much data to read or write, respectively.  Note,
	     these ioctls were added to aid in porting third party
	     applications and libraries, and should not be used in new code.

     AUDIO_GETINFO audio_info_t *
     AUDIO_SETINFO audio_info_t *
	     Get or set audio information as encoded in the audio_info

	     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_char  output_muted;   /* toggle play mute */
		     u_char  cspare[3];
		     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


	     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

	     blocksize is used to attempt to set both play and record block
	     sizes to the same value, it is left for compatibility only and
	     its use is discouraged.

	     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.

	     struct audio_prinfo {
		     u_int   sample_rate;    /* sample rate in bit/s */
		     u_int   channels;	     /* number of channels, usually 1 or 2 */
		     u_int   precision;	     /* number of bits/sample */
		     u_int   bps;	     /* number of bytes/sample */
		     u_int   msb;	     /* data alignment */
		     u_int   encoding;	     /* data encoding (AUDIO_ENCODING_* below) */
		     u_int   gain;	     /* volume level */
		     u_int   port;	     /* selected I/O port */
		     u_int   seek;	     /* BSD extension */
		     u_int   avail_ports;    /* available I/O ports */
		     u_int   buffer_size;    /* total size audio buffer */
		     u_int   block_size;     /* size a block */
		     /* 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
					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

	     The precision parameter describes the number of bits of audio
	     data per sample.  The bps parameter describes the number of bytes
	     of audio data per sample.	The msb parameter describes the
	     alignment of the data in the sample.  It is only meaningful when
	     precision / NBBY < bps.  A value of 1 means the data is aligned
	     to the most significant bit.

	     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] 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.

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

	     block_size sets the current audio block size.  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
	     block_size set is returned in this field.	Normally the
	     block_size is calculated to correspond to 50ms of sound and it is
	     recalculated when the encoding parameter changes, but if the
	     block_size 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 block_size to 0.

	     Care should be taken when setting the block_size before other
	     parameters.  If the device does not natively support the audio
	     parameters, then the internal block size may be scaled to a
	     larger size to accommodate conversion to a native format.	If the
	     block_size has been set, the internal block size will not be
	     rescaled when the parameters, and thus possibly the scaling
	     factor, change.  This can result in a block size much larger than
	     was originally requested.	It is recommended to set block_size at
	     the same time as, or after, all other parameters have been set.

	     The seek and samples fields are only used for AUDIO_GETINFO.
	     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.

     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

	     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

     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 used.

	     The mixer_class field identifies what class of control this is.
	     This value is set to the index value used to query the class
	     itself.  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
	     level 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 the 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, and 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


     aucat(1), audioctl(1), cdio(1), mixerctl(1), ioctl(2), ossaudio(3),
     sio_open(3), ac97(4), uaudio(4), audio(9)

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

OpenBSD 4.9			 July 15, 2010			   OpenBSD 4.9

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