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NAME
     audio_hdr - audio header structure

SYNOPSIS
     #include <multimedia/libaudio.h>

OVERVIEW
     Digital audio data represents a quantized  approximation  of
     an  analog  audio  signal,  which itself is directly propor-
     tional to the rapid changes of air pressure that we perceive
     as  sound.   Pulse  Code Modulation (PCM) refers to the sam-
     pling, at discrete intervals, of an audio waveform  to  pro-
     duce  a  (usually linear) quantization.  In order to achieve
     the best approximation of a  signal,  the  highest  possible
     sampling  frequency  and precision should be used.  However,
     it is often  impractical  to  process  and  store  the  huge
     amounts of data that such a practice yields.

     In lowering the overall data rate, a number  of  compromises
     must be made.  To begin with, the sampling frequency is gen-
     erally reduced to a level commensurate with the  quality  of
     the  audio  signal.  For speech and telephony, 8, 12, and 16
     kHz sampling rates are typically  used.   For  high-fidelity
     audio, 44.1 and 48 kHz sampling rates are predominant.

     Sampling precision may also be reduced to yield  lower  data
     rates.   Various  encoding schemes further compress the sig-
     nal.  Some of these, such as u-law and A-law companding, map
     the  linear  PCM  domain  into  a pseudo-logarithmic domain.
     Like PCM encoding, these schemes maintain a one-to-one rela-
     tionship  between  individual samples and waveform quantiza-
     tions.  Other encoding schemes,  such  as  Differential  and
     Adaptive  Differential  PCM, convolve several samples into a
     single sound unit that must be treated as an atomic entity.

     Multiple channels of audio data may be combined into a  sin-
     gle audio stream by interleaving individual sound units from
     each channel.  For each quantized period of  time,  the  set
     consisting  of  one  sound  unit  from each audio channel is
     known as a sample frame.

AUDIO HEADER
     For programming purposes, the  encoding  of  audio  data  is
     described  by  an  in-core  Audio_hdr structure.  This audio
     header is defined in <multimedia/audio_hdr.h> as follows:
          typedef struct {
               unsigned  sample_rate;        /* samples per second */
               unsigned  samples_per_unit;   /* samples per unit */
               unsigned  bytes_per_unit;          /* bytes per sample unit */
               unsigned  channels;      /* # of interleaved channels */
               unsigned  encoding;      /* data encoding format */
               unsigned  data_size;          /* length of data (advisory) */

          } Audio_hdr;

     The sample_rate field is an integer  representing  the  sam-
     pling frequency, in samples per second, of the audio data.

     The samples_per_unit and bytes_per_unit fields  describe  an
     individual  sound  unit.  For PCM and related encodings, the
     samples_per_unit field is 1, and  bytes_per_unit  gives  the
     precision.    For   encodings  that  group  several  samples
     together, bytes_per_unit reflects the size of an atomic unit
     of audio data.

     The channels field describes the number of interleaved audio
     channels.  The quantity
          channels * bytes_per_unit
     gives the size, in bytes, of each sample frame.

     These four values are used to compute the byte offset  given
     a particular time, and vice versa.  Specifically:
          seconds = offset / C
          offset = seconds * C
     where:
          C  =  (channels  *  bytes_per_unit  *  sample_rate)   /
          samples_per_unit

     The encoding field enumerates the  specific  data  encoding.
     The following values are defined:

     AUDIO_ENCODING_ULAW                ISDN u-law companding

     AUDIO_ENCODING_ALAW                ISDN A-law companding

     AUDIO_ENCODING_LINEAR              linear PCM 2's-complement
                                        integer (0-center)

     AUDIO_ENCODING_FLOAT               linear   IEEE   floating-
                                        point (-1. < x < +1.)

     Note that floating-point data is  normalized  to  _1.   Many
     signal-processing algorithms are simplified by such normali-
     zation since the result of the multiplication of two signals
     is  guaranteed  to  lie  within the acceptable range.  Thus,
     bounds-checking need only be performed when converting to or
     from integer representations.

     The data_size field contains the number of bytes in the data
     stream.   This  value  is advisory only, as this quantity is
     not always known in advance.  In  such  cases,  the  special
     value AUDIO_UNKNOWN_SIZE should be used.

SEE ALSO
     audio_convert(3),      audio_filehdr(3),      audio_misc(3),
     audio_ulaw2linear(3)

WARNINGS
     The manual pages, header files, and object  library  associ-
     ated  with /usr/demo/SOUND/libaudio.a are furnished on an as
     is basis as a preliminary interface to several useful  audio
     data-processing  capabilities.   A  future  release  of  the
     operating system may redefine the  syntax  or  semantics  of
     some of the functions described herein.

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