AUDIO(4) OpenBSD Programmer's Manual AUDIO(4)NAME
audio, mixer - device-independent audio driver layer
SYNOPSIS
audio* at ...
#include <sys/types.h>
#include <sys/ioctl.h>
#include <sys/audioio.h>
#include <string.h>
DESCRIPTION
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.
SAMPLING DEVICES
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:
AUDIO_FLUSH
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.
AUDIO_DRAIN
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.
AUDIO_GETPROPS int *
This command gets a bit set of hardware properties. If the
hardware has a certain property, the corresponding bit is set,
otherwise it is not. The properties can have the following
values:
AUDIO_PROP_FULLDUPLEX The device admits full-duplex operation.
AUDIO_PROP_MMAP The device can be used with mmap(2).
AUDIO_PROP_INDEPENDENT The device can set the playing and
recording encoding parameters
independently.
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
structure.
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
structure.
typedef struct audio_info {
struct audio_prinfo play; /* info for play (output) side */
struct audio_prinfo record; /* info for record (input) side */
u_int monitor_gain; /* input to output mix */
/* BSD extensions */
u_int blocksize; /* H/W read/write block size */
u_int hiwat; /* output high water mark */
u_int lowat; /* output low water mark */
u_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
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
playback.
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.
MIXER DEVICE
The mixer device, /dev/mixer, may be manipulated with ioctl(2) but does
not support read(2) or write(2). It supports the following ioctl(2)
commands:
AUDIO_GETDEV audio_device_t *
This command is the same as described above for the sampling
devices.
AUDIO_MIXER_READ mixer_ctrl_t *
AUDIO_MIXER_WRITE mixer_ctrl_t *
These commands read the current mixer state or set new mixer
state for the specified device dev. type identifies which type
of value is supplied in the mixer_ctrl_t * 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 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
values.
FILES
/dev/audio
/dev/audioctl
/dev/sound
/dev/mixer
SEE ALSOaucat(1), audioctl(1), cdio(1), mixerctl(1), ioctl(2), ossaudio(3),
sio_open(3), ac97(4), uaudio(4), audio(9)BUGS
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