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AUDIO(4) Device Drivers Manual AUDIO(4)

audio — device-independent audio driver layer

#include <sys/audioio.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.

/dev/sound and /dev/audio can be opened at 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.

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.

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 corresponds to the underlying audio device driver and device number. e.g In the case of an hdafg 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 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 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, 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:

This command will return the audio channel in use.
This command will select the audio channel for subsequent ioctl calls.
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.
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.
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.
This command suspends the calling process until all queued playback samples have been played by the hardware.
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;
The command returns the current setting of the full duplex mode.
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.

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.
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:

the device admits full duplex operation.
the device can be used with mmap(2).
the device can set the playing and recording encoding parameters independently.
the device is capable of audio playback.
the device is capable of audio capture.
 
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 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;
 
 
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:

mu-law encoding, 8 bits/sample
A-law encoding, 8 bits/sample
two's complement signed linear encoding with the platform byte order
unsigned linear encoding with the platform byte order
ADPCM encoding, 8 bits/sample
two's complement signed linear encoding with little endian byte order
two's complement signed linear encoding with big endian byte order
unsigned linear encoding with little endian byte order
unsigned linear encoding with big endian byte order
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:

to select microphone input.
to select line input.
to select CD input.

The output port should be a combination of:

to select speaker output.
to select headphone output.
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.

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:

This command is the same as described above for the sampling devices.
 
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.

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.

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

audioctl(1), mixerctl(1), ioctl(2), ossaudio(3), midi(4), radio(4), sysctl(8)

aria(4), ess(4), gus(4), guspnp(4), pas(4), sb(4), wss(4), ym(4)

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)

bba(4)

uaudio(4)

audio_system(9)

Support for virtual channels and mixing first appeared in NetBSD 8.0.

May 15, 2018 NetBSD-8.1