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 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 sample
mode and record/playback mode. In all other respects
/dev/audio and
/dev/sound are identical.
Only one process may hold open a sampling device at a given
time (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;
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) (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 *
This command fetches the count of dropped input samples into its
int * 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 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 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 halfduplex 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_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_int _ispare1;
u_int mode; /* current device
mode */
#define AUMODE_PLAY 0x01
#define AUMODE_RECORD 0x02
#define AUMODE_PLAY_ALL 0x04 /* do not do realtime correction */
} audio_info_t;
When setting the current state with AUDIO_SETINFO,
the audio_info
structure should first be initialized with
ioctl(fd, 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.
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 recalculated
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_int 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 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 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 blocksize gives the maximum value for hiwat.
Currently the buffer_size can only be read and not
set.
The seek and samples fields are only used for AUDIO_GETINFO.
seek represents the count of samples pending;
samples represents
the total number of bytes recorded or played, less
those that
were dropped due to inadequate consumption/production rates.
pause returns the current pause/unpause state for
recording or
playback. For AUDIO_SETINFO, if the pause value is
specified it
will either pause or unpause the particular direction.
The mixer device, /dev/mixer, may be manipulated with
ioctl(2) but does
not support read(2) or write(2). It supports the following
ioctl(2) 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.
/dev/audio
/dev/audioctl
/dev/sound
/dev/mixer
audioctl(1), mixerctl(1), ioctl(2), ossaudio(3), ac97(4),
uaudio(4),
audio(9)
For ports using the ISA bus: gus(4), pss(4), sb(4), wss(4)
For ports using the PCI bus: aria(4), auich(4), autri(4),
auvia(4),
clcs(4), clct(4), cmpci(4), eap(4), emu(4), esa(4), eso(4),
ess(4),
fms(4), maestro(4), neo(4), sv(4), yds(4), ym(4)
If the device is used in mmap(2) it is currently always
mapped for writing
(playing) due to VM system weirdness.
OpenBSD 3.6 March 11, 1997
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