AUDIO(4I) | Ioctl Requests | AUDIO(4I) |
audio
— generic
audio device interface
#include
<sys/audio.h>
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.
It is no longer possible to disable the mixing function. Applications must not assume that they have exclusive access to the audio device.
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.
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 μ-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 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.
An encoding parameter specifies the audio data representation. μ-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 μ-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 indicates the number of bits used to store each audio sample. For instance, μ-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.
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
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 |
μ-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.
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.
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_NDELAY
or
O_NONBLOCK
flags are set in the
open(2) oflag
argument, then -1 is immediately returned, with
errno set to EBUSY
.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 μ-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.
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, μ-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.
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 μ-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.
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.
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.
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:
AUDIO_SETINFO
ioctl has altered the device
state.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.
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.
The audio device additionally supports the following ioctl(2) commands:
AUDIO_DRAIN
AUDIO_DRAIN
is performed on the final
close(2) of
/dev/audio.AUDIO_GETDEV
AUDIO_GETINFO
AUDIO_SETINFO
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
.
The physical audio device names are system dependent and are rarely used by programmers. Programmers should use the following generic device names:
An open(2) call will fail if:
EBUSY
O_NDELAY
or O_NONBLOCK
flag was set in the open(2)
request.EINTR
An ioctl(2) call will fail if:
EINVAL
AUDIO_SETINFO
ioctl are invalid or are not
supported by the device.SPARC X86
Obsolete Uncommitted
close(2), fcntl(2), ioctl(2), open(2), poll(2), read(2), write(2), dsp(4I), streamio(4I), attributes(7)
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.
July 8, 2018 | OmniOS |