AUDIO(4I) Ioctl Requests AUDIO(4I)

NAME

audio - generic audio device interface

SYNOPSIS

#include <sys/audio.h>

OVERVIEW

An audio device is used to play and/or record a stream of audio data.
Since a specific audio device may not support all functionality
described below, refer to the device-specific manual pages for a
complete description of each hardware device. An application can use
the AUDIO_GETDEV ioctl(2) to determine the current audio hardware
associated with /dev/audio.

The audio framework provides a software mixing engine (audio mixer) for
all audio devices, allowing more than one process to play or record
audio at the same time.

Backward Compatibility

It is no longer possible to disable the mixing function. Applications
must not assume that they have exclusive access to the audio device.

Multi-Stream Codecs
The audio mixer supports multi-stream Codecs. These devices have DSP
engines that provide sample rate conversion, hardware mixing, and other
features. The use of such hardware features is opaque to applications.

AUDIO FORMATS

Digital audio data represents a quantized approximation of an analog
audio signal waveform. In the simplest case, these quantized numbers
represent the amplitude of the input waveform at particular sampling
intervals. To achieve the best approximation of an input signal, the
highest possible sampling frequency and precision should be used.
However, increased accuracy comes at a cost of increased data storage
requirements. For instance, one minute of monaural audio recorded in
<mu>-Law format (pronounced mew-law) at 8 KHz requires nearly 0.5
megabytes of storage, while the standard Compact Disc audio format
(stereo 16-bit linear PCM data sampled at 44.1 KHz) requires
approximately 10 megabytes per minute.

Audio data may be represented in several different formats. An audio
device's current audio data format can be determined by using the
AUDIO_GETINFO ioctl(2) described below.

An audio data format is characterized in the audio driver by four
parameters: Sample Rate, Encoding, Precision, and Channels. Refer to
the device-specific manual pages for a list of the audio formats that
each device supports. In addition to the formats that the audio device
supports directly, other formats provide higher data compression.
Applications may convert audio data to and from these formats when
playing or recording.

Sample Rate

Sample rate is a number that represents the sampling frequency (in
samples per second) of the audio data.

The audio mixer always configures the hardware for the highest possible
sample rate for both play and record. This ensures that none of the
audio streams require compute-intensive low pass filtering. The result
is that high sample rate audio streams are not degraded by filtering.

Sample rate conversion can be a compute-intensive operation, depending
on the number of channels and a device's sample rate. For example, an
8KHz signal can be easily converted to 48KHz, requiring a low cost up
sampling by 6. However, converting from 44.1KHz to 48KHz is compute
intensive because it must be up sampled by 160 and then down sampled by
147. This is only done using integer multipliers.

Applications can greatly reduce the impact of sample rate conversion by
carefully picking the sample rate. Applications should always use the
highest sample rate the device supports. An application can also do
its own sample rate conversion (to take advantage of floating point and
accelerated instruction or use small integers for up and down sampling.

All modern audio devices run at 48 kHz or a multiple thereof, hence
just using 48 kHz may be a reasonable compromise if the application is
not prepared to select higher sample rates.

Encodings

An encoding parameter specifies the audio data representation.
<mu>-Law encoding corresponds to CCITT G.711, and is the standard for
voice data used by telephone companies in the United States, Canada,
and Japan. A-Law encoding is also part of CCITT G.711 and is the
standard encoding for telephony elsewhere in the world. A-Law and
<mu>-Law audio data are sampled at a rate of 8000 samples per second
with 12-bit precision, with the data compressed to 8-bit samples. The
resulting audio data quality is equivalent to that of standard analog
telephone service.

Linear Pulse Code Modulation (PCM) is an uncompressed, signed audio
format in which sample values are directly proportional to audio signal
voltages. Each sample is a 2's complement number that represents a
positive or negative amplitude.

Precision

Precision indicates the number of bits used to store each audio sample.
For instance, <mu>-Law and A-Law data are stored with 8-bit precision.
PCM data may be stored at various precisions, though 16-bit is the most
common.

Channels

Multiple channels of audio may be interleaved at sample boundaries. A
sample frame consists of a single sample from each active channel. For
example, a sample frame of stereo 16-bit PCM data consists of two
16-bit samples, corresponding to the left and right channel data.

The audio mixer sets the hardware to the maximum number of channels
supported. If a mono signal is played or recorded, it is mixed on the
first two (usually the left and right) channels only. Silence is mixed
on the remaining channels

Supported Formats

The audio mixer supports the following audio formats:

Encoding Precision Channels
Signed Linear PCM 32-bit Mono or Stereo
Signed Linear PCM 16-bit Mono or Stereo
Signed Linear PCM 8-bit Mono or Stereo
<mu>-Law 8-bit Mono or Stereo
A-Law 8-bit Mono or Stereo

The audio mixer converts all audio streams to 24-bit Linear PCM before
mixing. After mixing, conversion is made to the best possible Codec
format. The conversion process is not compute intensive and audio
applications can choose the encoding format that best meets their
needs.

Note that the mixer discards the low order 8 bits of 32-bit Signed
Linear PCM in order to perform mixing. (This is done to allow for
possible overflows to fit into 32-bits when mixing multiple streams
together.) Hence, the maximum effective precision is 24-bits.

DESCRIPTION

The device /dev/audio is a device driver that dispatches audio requests
to the appropriate underlying audio hardware. The audio driver is
implemented as a STREAMS driver. In order to record audio input,
applications open(2) the /dev/audio device and read data from it using
the read(2) system call. Similarly, sound data is queued to the audio
output port by using the write(2) system call. Device configuration is
performed using the ioctl(2) interface.

Because some systems may contain more than one audio device,
application writers are encouraged to query the AUDIODEV environment
variable. If this variable is present in the environment, its value
should identify the path name of the default audio device.

Opening the Audio Device

The audio device is not treated as an exclusive resource. Each process
may open the audio device once.

Each open(2) completes as long as there are channels available to be
allocated. If no channels are available to be allocated:

+o if either the O_NDELAY or O_NONBLOCK flags are set in the
open(2) oflag argument, then -1 is immediately returned, with
errno set to EBUSY.

+o if neither the O_NDELAY nor the O_NONBLOCK flag are set, then
open(2) hangs until the device is available or a signal is
delivered to the process, in which case a -1 is returned with
errno set to EINTR.

Upon the initial open(2) of the audio channel, the audio mixer sets the
data format of the audio channel to the default state of 8-bit, 8Khz,
mono <mu>-Law data.

If the audio device does not support this configuration, it informs the
audio mixer of the initial configuration. Audio applications should
explicitly set the encoding characteristics to match the audio data
requirements, and not depend on the default configuration.

Recording Audio Data

The read(2) system call copies data from the system's buffers to the
application. Ordinarily, read(2) blocks until the user buffer is
filled. The I_NREAD ioctl (see streamio(4I)) may be used to determine
the amount of data that may be read without blocking. The device may
alternatively be set to a non-blocking mode, in which case read(2)
completes immediately, but may return fewer bytes than requested.
Refer to the read(2) manual page for a complete description of this
behavior.

When the audio device is opened with read access, the device driver
immediately starts buffering audio input data. Since this consumes
system resources, processes that do not record audio data should open
the device write-only (O_WRONLY).

The transfer of input data to STREAMS buffers may be paused (or
resumed) by using the AUDIO_SETINFO ioctl to set (or clear) the
record.pause flag in the audio information structure (see below). All
unread input data in the STREAMS queue may be discarded by using the
I_FLUSH STREAMS ioctl. See streamio(4I). When changing record
parameters, the input stream should be paused and flushed before the
change, and resumed afterward. Otherwise, subsequent reads may return
samples in the old format followed by samples in the new format. This
is particularly important when new parameters result in a changed
sample size.

Input data can accumulate in STREAMS buffers very quickly. At a
minimum, it will accumulate at 8000 bytes per second for 8-bit, 8 KHz,
mono, <mu>-Law data. If the device is configured for 16-bit linear or
higher sample rates, it will accumulate even faster. If the
application that consumes the data cannot keep up with this data rate,
the STREAMS queue may become full. When this occurs, the record.error
flag is set in the audio information structure and input sampling
ceases until there is room in the input queue for additional data. In
such cases, the input data stream contains a discontinuity. For this
reason, audio recording applications should open the audio device when
they are prepared to begin reading data, rather than at the start of
extensive initialization.

Playing Audio Data

The write(2) system call copies data from an application's buffer to
the STREAMS output queue. Ordinarily, write(2) blocks until the entire
user buffer is transferred. The device may alternatively be set to a
non-blocking mode, in which case write(2) completes immediately, but
may have transferred fewer bytes than requested. See write(2).

Although write(2) returns when the data is successfully queued, the
actual completion of audio output may take considerably longer. The
AUDIO_DRAIN ioctl may be issued to allow an application to block until
all of the queued output data has been played. Alternatively, a
process may request asynchronous notification of output completion by
writing a zero-length buffer (end-of-file record) to the output stream.
When such a buffer has been processed, the play.eof flag in the audio
information structure is incremented.

The final close(2) of the file descriptor hangs until all of the audio
output has drained. If a signal interrupts the close(2), or if the
process exits without closing the device, any remaining data queued for
audio output is flushed and the device is closed immediately.

The consumption of output data may be paused (or resumed) by using the
AUDIO_SETINFO ioctl to set (or clear) the play.pause flag in the audio
information structure. Queued output data may be discarded by using
the I_FLUSH STREAMS ioctl. (See streamio(4I)).

Output data is played from the STREAMS buffers at a default rate of at
least 8000 bytes per second for <mu>-Law, A-Law or 8-bit PCM data
(faster for 16-bit linear data or higher sampling rates). If the
output queue becomes empty, the play.error flag is set in the audio
information structure and output is stopped until additional data is
written. If an application attempts to write a number of bytes that is
not a multiple of the current sample frame size, an error is generated
and the bad data is thrown away. Additional writes are allowed.

Asynchronous I/O
The I_SETSIG STREAMS ioctl enables asynchronous notification, through
the SIGPOLL signal, of input and output ready condition changes. The
O_NONBLOCK flag may be set using the F_SETFL fcntl(2) to enable non-
blocking read(2) and write(2) requests. This is normally sufficient
for applications to maintain an audio stream in the background.

Audio Control Pseudo-Device
It is sometimes convenient to have an application, such as a volume
control panel, modify certain characteristics of the audio device while
it is being used by an unrelated process.

The /dev/audioctl pseudo-device is provided for this purpose. Any
number of processes may open /dev/audioctl simultaneously. However,
read(2) and write(2) system calls are ignored by /dev/audioctl. The
AUDIO_GETINFO and AUDIO_SETINFO ioctl commands may be issued to
/dev/audioctl to determine the status or alter the behavior of
/dev/audio. Note: In general, the audio control device name is
constructed by appending the letters "ctl" to the path name of the
audio device.

Audio Status Change Notification

Applications that open the audio control pseudo-device may request
asynchronous notification of changes in the state of the audio device
by setting the S_MSG flag in an I_SETSIG STREAMS ioctl. Such processes
receive a SIGPOLL signal when any of the following events occur:

+o An AUDIO_SETINFO ioctl has altered the device state.

+o An input overflow or output underflow has occurred.

+o An end-of-file record (zero-length buffer) has been processed
on output.

+o An open(2) or close(2) of /dev/audio has altered the device
state.

+o An external event (such as speakerbox's volume control) has
altered the device state.

IOCTLS

Audio Information Structure

The state of the audio device may be polled or modified using the
AUDIO_GETINFO and AUDIO_SETINFO ioctl commands. These commands operate
on the audio_info structure as defined, in <sys/audio.h>, as follows:

/*
* This structure contains state information for audio device
* IO streams
*/

struct audio_prinfo {
/*
* The following values describe the
* audio data encoding
*/
uint_t sample_rate; /* samples per second */
uint_t channels; /* number of interleaved channels */
uint_t precision; /* number of bits per sample */
uint_t encoding; /* data encoding method */

/*
* The following values control audio device
* configuration
*/
uint_t gain; /* volume level */
uint_t port; /* selected I/O port */
uint_t buffer_size; /* I/O buffer size */

/*
* The following values describe the current device
* state
*/
uint_t samples; /* number of samples converted */
uint_t eof; /* End Of File counter (play only) */
uchar_t pause; /* non-zero if paused, zero to resume */
uchar_t error; /* non-zero if overflow/underflow */
uchar_t waiting; /* non-zero if a process wants access */
uchar_t balance; /* stereo channel balance */

/*
* The following values are read-only device state
* information
*/
uchar_t open; /* non-zero if open access granted */
uchar_t active; /* non-zero if I/O active */
uint_t avail_ports; /* available I/O ports */
uint_t mod_ports; /* modifiable I/O ports */
};
typedef struct audio_prinfo audio_prinfo_t;

/*
* This structure is used in AUDIO_GETINFO and AUDIO_SETINFO ioctl
* commands
*/
struct audio_info {
audio_prinfo_t record;/* input status info */
audio_prinfo_t play; /* output status info */
uint_t monitor_gain; /* input to output mix */
uchar_toutput_muted; /* non-zero if output muted */
uint_t hw_features; /* supported H/W features */
uint_t sw_features; /* supported S/W features */
uint_t sw_features_enabled;
/* supported S/W features enabled */
};
typedef struct audio_info audio_info_t;

/* Audio encoding types */
#define AUDIO_ENCODING_ULAW (1) /* u-Law encoding */
#define AUDIO_ENCODING_ALAW (2) /* A-Law encoding */
#define AUDIO_ENCODING_LINEAR (3) /* Signed Linear PCM encoding */

/*
* These ranges apply to record, play, and
* monitor gain values
*/
#define AUDIO_MIN_GAIN (0)/* minimum gain value */
#define AUDIO_MAX_GAIN (255) /* maximum gain value */

/*
* These values apply to the balance field to adjust channel
* gain values
*/
#define AUDIO_LEFT_BALANCE (0) /* left channel only */
#define AUDIO_MID_BALANCE (32) /* equal left/right balance */
#define AUDIO_RIGHT_BALANCE (64) /* right channel only */

/*
* Define some convenient audio port names
* (for port, avail_ports and mod_ports)
*/

/* output ports (several might be enabled at once) */
#define AUDIO_SPEAKER (0x01) /* built-in speaker */
#define AUDIO_HEADPHONE (0x02) /* headphone jack */
#define AUDIO_LINE_OUT (0x04) /* line out */
#define AUDIO_SPDIF_OUT (0x08) /* SPDIF port */
#define AUDIO_AUX1_OUT (0x10) /* aux1 out */
#define AUDIO_AUX2_OUT (0x20) /* aux2 out */

/*
* input ports (usually only one may be enabled at a time)
*/
#define AUDIO_MICROPHONE (0x01) /* microphone */
#define AUDIO_LINE_IN (0x02) /* line in */
#define AUDIO_CD (0x04) /* on-board CD inputs */
#define AUDIO_SPDIF_IN (0x08) /* SPDIF port */
#define AUDIO_AUX1_IN (0x10) /* aux1 in */
#define AUDIO_AUX2_IN (0x20) /* aux2 in */
#define AUDIO_CODEC_LOOPB_IN (0x40) /* Codec inter. loopback */

/* These defines are for hardware features */
#define AUDIO_HWFEATURE_DUPLEX (0x00000001u)
/* simult. play & cap. supported */

#define AUDIO_HWFEATURE_MSCODEC (0x00000002u)
/* multi-stream Codec */

/* These defines are for software features *
#define AUDIO_SWFEATURE_MIXER (0x00000001u)
/* audio mixer audio pers. mod. */

/*
* Parameter for the AUDIO_GETDEV ioctl
* to determine current audio devices
*/
#define MAX_AUDIO_DEV_LEN (16)
struct audio_device {
char name[MAX_AUDIO_DEV_LEN];
char version[MAX_AUDIO_DEV_LEN];
char config[MAX_AUDIO_DEV_LEN];
};
typedef struct audio_device audio_device_t;

The play.gain and record.gain fields specify the output and input
volume levels. A value of AUDIO_MAX_GAIN indicates maximum volume.
Audio output may also be temporarily muted by setting a non-zero value
in the output_muted field. Clearing this field restores audio output
to the normal state.

The monitor_gain field is present for compatibility, and is no longer
supported. See dsp(4I) for more detail.

Likewise, the play.port, play.ports, play.mod_ports, record.port,
record.ports, and record.mod_ports are no longer supported. See
dsp(4I) for more detail.

The play.balance and record.balance fields are fixed to
AUDIO_MID_BALANCE. Changes to volume levels for different channels can
be made using the interfaces in dsp(4I).

The play.pause and record.pause flags may be used to pause and resume
the transfer of data between the audio device and the STREAMS buffers.
The play.error and record.error flags indicate that data underflow or
overflow has occurred. The play.active and record.active flags
indicate that data transfer is currently active in the corresponding
direction.

The play.open and record.open flags indicate that the device is
currently open with the corresponding access permission. The
play.waiting and record.waiting flags provide an indication that a
process may be waiting to access the device. These flags are set
automatically when a process blocks on open(2), though they may also be
set using the AUDIO_SETINFO ioctl command. They are cleared only when
a process relinquishes access by closing the device.

The play.samples and record.samples fields are zeroed at open(2) and
are incremented each time a data sample is copied to or from the
associated STREAMS queue. Some audio drivers may be limited to
counting buffers of samples, instead of single samples for their
samples accounting. For this reason, applications should not assume
that the samples fields contain a perfectly accurate count. The
play.eof field increments whenever a zero-length output buffer is
synchronously processed. Applications may use this field to detect the
completion of particular segments of audio output.

The record.buffer_size field controls the amount of input data that is
buffered in the device driver during record operations. Applications
that have particular requirements for low latency should set the value
appropriately. Note however that smaller input buffer sizes may result
in higher system overhead. The value of this field is specified in
bytes and drivers will constrain it to be a multiple of the current
sample frame size. Some drivers may place other requirements on the
value of this field. Refer to the audio device-specific manual page
for more details. If an application changes the format of the audio
device and does not modify the record.buffer_size field, the device
driver may use a default value to compensate for the new data rate.
Therefore, if an application is going to modify this field, it should
modify it during or after the format change itself, not before. When
changing the record.buffer_size parameters, the input stream should be
paused and flushed before the change, and resumed afterward.
Otherwise, subsequent reads may return samples in the old format
followed by samples in the new format. This is particularly important
when new parameters result in a changed sample size. If you change the
record.buffer_size for the first packet, this protocol must be followed
or the first buffer will be the default buffer size for the device,
followed by packets of the requested change size.

The record.buffer_size field may be modified only on the /dev/audio
device by processes that have it opened for reading.

The play.buffer_size field is currently not supported.

The audio data format is indicated by the sample_rate, channels,
precision, and encoding fields. The values of these fields correspond
to the descriptions in the AUDIO FORMATS section of this man page.
Refer to the audio device-specific manual pages for a list of supported
data format combinations.

The data format fields can be modified only on the /dev/audio device.

If the parameter changes requested by an AUDIO_SETINFO ioctl cannot all
be accommodated, ioctl(2) returns with errno set to EINVAL and no
changes are made to the device state.

Streamio IOCTLS

All of the streamio(4I) ioctl(2) commands may be issued for the
/dev/audio device. Because the /dev/audioctl device has its own
STREAMS queues, most of these commands neither modify nor report the
state of /dev/audio if issued for the /dev/audioctl device. The
I_SETSIG ioctl may be issued for /dev/audioctl to enable the
notification of audio status changes, as described above.

Audio IOCTLS

The audio device additionally supports the following ioctl(2) commands:

AUDIO_DRAIN The argument is ignored. This command suspends the
calling process until the output STREAMS queue is empty
and all queued samples have been played, or until a
signal is delivered to the calling process. It may not
be issued for the /dev/audioctl device. An implicit
AUDIO_DRAIN is performed on the final close(2) of
/dev/audio.

AUDIO_GETDEV The argument is a pointer to an audio_device_t
structure. This command may be issued for either
/dev/audio or /dev/audioctl. The returned value in the
name field will be a string that will identify the
current /dev/audio hardware device, the value in version
will be a string indicating the current version of the
hardware, and config will be a device-specific string
identifying the properties of the audio stream
associated with that file descriptor. Refer to the
audio device-specific manual pages to determine the
actual strings returned by the device driver.

AUDIO_GETINFO The argument is a pointer to an audio_info_t structure.
This command may be issued for either /dev/audio or
/dev/audioctl. The current state of the /dev/audio
device is returned in the structure. Values return
pertain to a logical view of the device as seen by and
private to the process, and do not necessarily reflect
the actual hardware device itself.

AUDIO_SETINFO The argument is a pointer to an audio_info_t structure.
This command may be issued for either the /dev/audio or
the /dev/audioctl device with some restrictions. This
command configures the audio device according to the
supplied structure and overwrites the existing structure
with the new state of the device. Note: The
play.samples, record.samples, play.error, record.error,
and play.eof fields are modified to reflect the state of
the device when the AUDIO_SETINFO is issued. This
allows programs to automatically modify these fields
while retrieving the previous value.

As with AUDIO_SETINFO, the settings managed by this
ioctl deal with a logical view of the device which is
private to the process, and don't necessarily have any
impact on the hardware device itself.

Certain fields in the audio information structure, such as the pause
flags, are treated as read-only when /dev/audio is not open with the
corresponding access permission. Other fields, such as the gain levels
and encoding information, may have a restricted set of acceptable
values. Applications that attempt to modify such fields should check
the returned values to be sure that the corresponding change took
effect. The sample_rate, channels, precision, and encoding fields
treated as read-only for /dev/audioctl, so that applications can be
guaranteed that the existing audio format will stay in place until they
relinquish the audio device. AUDIO_SETINFO will return EINVAL when the
desired configuration is not possible, or EBUSY when another process
has control of the audio device.

All of the logical device state is reset when the corresponding I/O
stream of /dev/audio is closed.

The audio_info_t structure may be initialized through the use of the
AUDIO_INITINFO macro. This macro sets all fields in the structure to
values that are ignored by the AUDIO_SETINFO command. For instance,
the following code switches the output port from the built-in speaker
to the headphone jack without modifying any other audio parameters:

audio_info_t info;
AUDIO_INITINFO();
info.play.port = AUDIO_HEADPHONE;
err = ioctl(audio_fd, AUDIO_SETINFO, );

This technique eliminates problems associated with using a sequence of
AUDIO_GETINFO followed by AUDIO_SETINFO.

FILES

The physical audio device names are system dependent and are rarely
used by programmers. Programmers should use the following generic
device names:

/dev/audio Symbolic link to the system's primary audio
device

/dev/audioctl Symbolic link to the control device for
/dev/audio

/dev/sound/0 First audio device in the system

/dev/sound/0ctl Audio control device for /dev/sound/0

/usr/share/audio/samples Audio files

ERRORS

An open(2) call will fail if:

EBUSY The requested play or record access is busy and either the
O_NDELAY or O_NONBLOCK flag was set in the open(2) request.

EINTR The requested play or record access is busy and a signal
interrupted the open(2) request.

An ioctl(2) call will fail if:

EINVAL The parameter changes requested in the AUDIO_SETINFO ioctl are
invalid or are not supported by the device.

ARCHITECTURE

SPARC X86

INTERFACE STABILITY

Obsolete Uncommitted

SEE ALSO

close(2), fcntl(2), ioctl(2), open(2), poll(2), read(2), write(2),
dsp(4I), streamio(4I), attributes(7)

BUGS

Due to a feature of the STREAMS implementation, programs that are
terminated or exit without closing the audio device may hang for a
short period while audio output drains. In general, programs that
produce audio output should catch the SIGINT signal and flush the
output stream before exiting.

illumos July 8, 2018 illumos