AUDIO(4) | Device Drivers Manual | AUDIO(4) |
audio
—
#include <sys/audioio.h>
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.
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:
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:
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)
AUDIO_SETCHAN
(int)
AUDIO_FLUSH
AUDIO_PERROR
(int)
AUDIO_RERROR
(int)
AUDIO_WSEEK
(u_long)
AUDIO_DRAIN
AUDIO_GETDEV
(audio_device_t)
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_GETENC
(audio_encoding_t)
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_GETFD
(int)
AUDIO_SETFD
(int)
AUDIO_GETPROPS
(int)
AUDIO_PROP_FULLDUPLEX
AUDIO_PROP_MMAP
AUDIO_PROP_INDEPENDENT
AUDIO_PROP_PLAYBACK
AUDIO_PROP_CAPTURE
AUDIO_GETIOFFS
(audio_offset_t)
AUDIO_GETOOFFS
(audio_offset_t)
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)
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
AUDIO_ENCODING_ALAW
AUDIO_ENCODING_SLINEAR
AUDIO_ENCODING_ULINEAR
AUDIO_ENCODING_ADPCM
AUDIO_ENCODING_SLINEAR_LE
AUDIO_ENCODING_SLINEAR_BE
AUDIO_ENCODING_ULINEAR_LE
AUDIO_ENCODING_ULINEAR_BE
AUDIO_ENCODING_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
AUDIO_LINE_IN
AUDIO_CD
The output port should be a combination of:
AUDIO_SPEAKER
AUDIO_HEADPHONE
AUDIO_LINE_OUT
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.
AUDIO_GETDEV
(audio_device_t)
AUDIO_MIXER_READ
(mixer_ctrl_t)
AUDIO_MIXER_WRITE
(mixer_ctrl_t)
#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)
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.
May 15, 2018 | NetBSD 8.99 |