NAME
audio —
device-independent audio driver
layer
SYNOPSIS
#include <sys/audioio.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.
/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
sample 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 identical.
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 encoding,
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
hdaudio(4) supported device the
variables would be:
hw.hdafg0.channels
,
hw.hdafg0.precision
,
hw.hdafg0.frequency
.
For best results, values close to the underlying hardware should be chosen.
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.dac
N
- Output volume
vchan.mic
N
- Recording volume
Where
N is the virtual channel number. E.g.
vchan.dac0
controlling playback 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 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) 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(2), 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 playback
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
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)
- 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 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_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(getting) 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_t 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
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 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
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_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 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
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/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. 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 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 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
,
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 ALSO
audioctl(1),
mixerctl(1),
ioctl(2),
ossaudio(3),
midi(4),
radio(4),
sysctl(8)
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)
TURBOchannel
bba(4)
USB
uaudio(4)
HISTORY
Support for virtual channels and mixing first appeared in
NetBSD 8.0.