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11641 spelling mistakes in section 7d of the manual
@@ -1,16 +1,14 @@
'\" te
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-.TH AUDIO 7D "Aug 3, 2009"
+.TH AUDIO 7D "Jan 10, 2020"
.SH NAME
audio \- common audio framework
.SH DESCRIPTION
-.sp
-.LP
The \fBaudio\fR driver provides common support routines for audio devices in
Solaris.
.sp
.LP
The audio framework supports multiple \fBpersonalities\fR, allowing for devices
@@ -18,29 +16,21 @@
.sp
.LP
The audio framework also provides a number of facilities, such as mixing of
audio streams, and data format and sample rate conversion.
.SS "Overview"
-.sp
-.LP
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.
.SS "Multi-Stream Codecs"
-.sp
-.LP
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.
.SS "Backward Compatibility"
-.sp
-.LP
It is not possible to disable the mixing function. Applications must not assume
that they have exclusive access to the audio device.
.SS "Audio Formats"
-.sp
-.LP
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
@@ -56,12 +46,10 @@
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.
.SS "Sample Rate"
-.sp
-.LP
Sample rate is a number that represents the sampling frequency (in samples per
second) of the audio data.
.sp
.LP
The audio mixer always configures the hardware for the highest possible sample
@@ -68,11 +56,11 @@
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.
.sp
.LP
-Sample rate conversion can be a compute-intensive operation, dependingon the
+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.
@@ -87,12 +75,10 @@
.LP
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.
.SS "Encodings"
-.sp
-.LP
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
@@ -104,28 +90,22 @@
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.
.SS "Precision"
-.sp
-.LP
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.
.SS "Channels"
-.sp
-.LP
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.
.SS "Supported Formats"
-.sp
-.LP
The audio mixer supports the following audio formats:
.sp
.in +2
.nf
Encoding Precision Channels
@@ -149,49 +129,37 @@
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.
.SH FILES
-.sp
.ne 2
.na
-\fB\fB/kernel/drv/audio\fR\fR
-.ad
-.RS 29n
-32-bit kernel driver module
-.RE
-
-.sp
-.ne 2
-.na
\fB\fB/kernel/drv/amd64/audio\fR\fR
.ad
.RS 29n
-64-bit x86 kernel driver module
+Device driver (x86)
.RE
.sp
.ne 2
.na
\fB\fB/kernel/drv/sparcv9/audio\fR\fR
.ad
.RS 29n
-64-bit SPARC kernel driver module
+Device driver (SPARC)
.RE
.sp
.ne 2
.na
\fB\fB/kernel/drv/audio.conf\fR\fR
.ad
.RS 29n
-\fBaudio\fR configuration file
+Driver configuration file
.RE
.SH ATTRIBUTES
-.sp
-.LP
See \fBattributes\fR(5) for a description of the following attributes:
.sp
.sp
.TS
@@ -204,8 +172,6 @@
_
Interface Stability Uncommitted
.TE
.SH SEE ALSO
-.sp
-.LP
\fBioctl\fR(2), \fBattributes\fR(5), \fBaudio\fR(7I), \fBdsp\fR(7I)