illumos: manual page: audio.4d

AUDIO(4D) AUDIO(4D)

NAME

audio - common audio framework

DESCRIPTION

The audio driver provides common support routines for audio devices in
Solaris.

The audio framework supports multiple personalities, allowing for devices
to be accessed with different programming interfaces.

The audio framework also provides a number of facilities, such as mixing
of audio streams, and data format and sample rate conversion.

Overview

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.

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.

Backward Compatibility

It is not possible to disable the mixing function. Applications must not
assume that they have exclusive access to the audio device.

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 u-
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.

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 can 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 computer 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 instructions) 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 can be a reasonable compromise if the application is not
prepared to select higher sample rates.

Encodings

An encoding parameter specifies the audiodata representation. u-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 u-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 stan dard 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, u-Law and A-Law data are stored with 8-bit precision. PCM
data can be stored at various precisions, though 16-bit is the most
common.

Channels

Multiple channels of audio can 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 2 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) channel 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
u-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.

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.

FILES

/kernel/drv/amd64/audio
Device driver (x86)

/kernel/drv/sparcv9/audio
Device driver (SPARC)

/kernel/drv/audio.conf
Driver configuration file

ATTRIBUTES