Manual Pages for UNIX Operating System command usage for man audio

Devices                                                 audio(7D)



NAME
     audio - common audio framework

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


     The audio framework supports multiple personalities,  allow-
     ing  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.  Applica-
     tions 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 chan-
     nels. Refer to the device-specific manual pages for  a  list



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Devices                                                 audio(7D)



     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  play-
     ing or recording.

  Sample Rate
     Sample rate is a number that represents  the  sampling  fre-
     quency (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,
     dependingon  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, con-
     verting 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. Applica-
     tions 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  representa-
     tion.  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.



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Devices                                                 audio(7D)



     Linear Pulse  Code  Modulation  (PCM)  is  an  uncompressed,
     signed audio format in which sample values are directly pro-
     portional 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 pre-
     cisions, 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/audio            32-bit kernel driver module




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Devices                                                 audio(7D)



     /kernel/drv/amd64/audio      64-bit x86 kernel driver module


     /kernel/drv/sparcv9/audio    64-bit  SPARC   kernel   driver
                                  module


     /kernel/drv/audio.conf       audio configuration file


ATTRIBUTES
     See attributes(5) for a description of the following  attri-
     butes:



     ____________________________________________________________
    | ATTRIBUTE TYPE              | ATTRIBUTE VALUE             |
    |_____________________________|_____________________________|
    | Architecture                | SPARC, x86                  |
    |_____________________________|_____________________________|
    | Availability                | driver/audio                |
    |_____________________________|_____________________________|
    | Interface Stability         | Uncommitted                 |
    |_____________________________|_____________________________|


SEE ALSO
     ioctl(2), attributes(5), audio(7I), dsp(7I)


























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