audio(4) - NetBSD Manual Pages

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AUDIO(4) NetBSD Kernel Interfaces Manual AUDIO(4)

NAME

audio -- device-independent audio driver layer

SYNOPSIS

#include <sys/audioio.h>

DESCRIPTION

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
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.

/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
use.

SAMPLING DEVICES

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-
tical.

VIRTUAL CHANNELS

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.

hw.driverN.precision

hw.driverN.frequency

hw.driverN.channels

hw.driverN.latency

hw.driverN.multiuser

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-
quency.

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
have.

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
variable.

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
device.

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
calls.

AUDIO_FLUSH
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.

AUDIO_DRAIN
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.

AUDIO_GETPROPS (int)
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-
dently.
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-
ture.

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
AUDIO_GETINFO or AUDIO_GETBUFINFO 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 play-
back.

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
only).

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.

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) com-
mands:

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 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
used.

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
values.

FILES

/dev/audio
/dev/audioctl
/dev/sound
/dev/mixer