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audioconvert(1)                  User Commands                 audioconvert(1)

       audioconvert - convert audio file formats

       audioconvert [-pF] [-f outfmt] [-o outfile] [ [-i infmt] [file...]] ...

       audioconvert  converts  audio  data  between  a  set of supported audio
       encodings and file formats. It can be used to compress  and  decompress
       audio  data,  to add audio file headers to raw audio data files, and to
       convert between standard data encodings, such as -law and linear PCM.

       If no filenames are present, audioconvert reads the data from the stan-
       dard input stream and writes an audio file to the standard output. Oth-
       erwise, input files are processed in order, concatenated,  and  written
       to the output file.

       Input  files  are  expected to contain audio file headers that identify
       the audio data format.  If the audio data does not contain a  recogniz-
       able header, the format must be specified with the -i option, using the
       rate, encoding, and channels keywords to identify the input  data  for-

       The  output  file format is derived by updating the format of the first
       input file with the format options in the -f specification.  If  -p  is
       not specified, all subsequent input files are converted to this result-
       ing format and concatenated together. The output file will  contain  an
       audio  file header, unless format=raw is specified in the output format

       Input files may be converted in place by using the -p option.  When  -p
       is  in  effect,  the format of each input file is modified according to
       the -f option to determine the output format. The  existing  files  are
       then overwritten with the converted data.

       The  file(1)  command  decodes  and prints the audio data format of Sun
       audio files.

       The following options are supported:

       -p              In Place: The input files are individually converted to
                       the format specified by the -f option and rewritten. If
                       a target file is a symbolic link, the  underlying  file
                       will  be  rewritten. The -o option may not be specified
                       with -p.

       -F              Force: This option forces audioconvert  to  ignore  any
                       file  header  for input files whose format is specified
                       by the -i option. If -F is not specified,  audioconvert
                       ignores  the  -i  option  for  input files that contain
                       valid audio file headers.

       -f outfmt       Output Format: This option is used to specify the  file
                       format  and  data encoding of the output file. Defaults
                       for unspecified fields are derived from the input  file
                       format.  Valid  keywords  and  values are listed in the
                       next section.

       -o outfile      Output File: All input  files  are  concatenated,  con-
                       verted  to  the output format, and written to the named
                       output file. If -o and -p are not specified,  the  con-
                       catenated output is written to the standard output. The
                       -p option may not be specified with -o.

       -i infmt        Input Format: This option is used to specify  the  data
                       encoding of raw input files. Ordinarily, the input data
                       format is derived from  the  audio  file  header.  This
                       option  is  required when converting audio data that is
                       not preceded by a valid audio file  header.  If  -i  is
                       specified for an input file that contains an audio file
                       header, the input format string will be ignored, unless
                       -F  is  present. The format specification syntax is the
                       same as the -f output file format.

                       Multiple input formats may be specified. An input  for-
                       mat describes all input files following that specifica-
                       tion, until a new input format is specified.

       file            File Specification: The named audio files are  concate-
                       nated, converted to the output format, and written out.
                       If no file name is present, or if the special file name
                       `-'  is specified, audio data is read from the standard

       -?              Help: Prints a command line usage message.

   Format Specification
       The syntax for the input and output format specification is:

              keyword=value[,keyword=value ...]

       with no intervening whitespace. Unambiguous values may be used  without
       the preceding keyword=.

       rate            The  audio  sampling  rate  is specified in samples per
                       second. If a number is followed by the letter k, it  is
                       multiplied  by 1000 (for example, 44.1k = 44100). Stan-
                       dard of the commonly used sample rates  are:  8k,  16k,
                       32k, 44.1k, and 48k.

       channels        The  number  of interleaved channels is specified as an
                       integer. The words mono and stereo may also be used  to
                       specify one and two channel data, respectively.

       encoding        This  option specifies the digital audio data represen-
                       tation. Encodings determine precision implicitly  (ulaw
                       implies  8-bit  precision) or explicitly as part of the
                       name (for example,  linear16).  Valid  encoding  values

                       ulaw            CCITT  G.711  -law encoding. This is an
                                       8-bit format primarily used  for  tele-
                                       phone quality speech.

                       alaw            CCITT  G.711 A-law encoding. This is an
                                       8-bit format primarily used  for  tele-
                                       phone quality speech in Europe.

                       linear8,        Linear   Pulse  Code  Modulation  (PCM)
                       linear16,       encoding. The name identifies the  num-
                       linear32        ber  of  bits of precision. linear16 is
                                       typically used for high  quality  audio

                       pcm             Same as linear16.

                       g721            CCITT  G.721  compression  format. This
                                       encoding uses Adaptive Delta Pulse Code
                                       Modulation  (ADPCM)  with  4-bit preci-
                                       sion. It is  primarily  used  for  com-
                                       pressing  -law  voice data (achieving a
                                       2:1 compression ratio).

                       g723            CCITT G.723  compression  format.  This
                                       encoding uses Adaptive Delta Pulse Code
                                       Modulation (ADPCM)  with  3-bit  preci-
                                       sion.  It  is  primarily  used for com-
                                       pressing -law voice data (achieving  an
                                       8:3 compression ratio). The audio qual-
                                       ity is similar to G.721, but may result
                                       in  lower  quality  when  used for non-
                                       speech data.

                       The following encoding  values  are  also  accepted  as
                       shorthand  to set the sample rate, channels, and encod-

                       voice    Equivalent   to    encoding=ulaw,rate=8k,chan-

                       cd       Equivalent           to          encoding=lin-

                       dat      Equivalent to encoding=linear16,rate=48k,chan-

       format          This option specifies the audio file format. Valid for-
                       mats are:

                       sun      Sun compatible file format (the default).

                       raw      Use this format when reading  or  writing  raw
                                audio  data (with no audio header), or in con-
                                junction with an  offset to import  a  foreign
                                audio file format.

       offset          (-i  only)  Specifies a byte offset to locate the start
                       of the audio data. This option may be  used  to  import
                       audio data that contains an unrecognized file header.

       See  largefile(5)  for  the description of the behavior of audioconvert
       when encountering files greater than or equal  to  2  Gbyte  (  2 **31

       Example 1: Recording and compressing voice data before storing it

       Record voice data and compress it before storing it to a file:

       example% audiorecord | audioconvert -f g721 >> mydata.au

       Example 2: Concatenating two audio files

       Concatenate  two  Sun format audio files, regardless of their data for-
       mat, and output an 8-bit ulaw, 16 kHz, mono file:

       example% audioconvert -f ulaw,rate=16k,mono -o outfile.au infile1 infile2

       Example 3: Converting a directory to Sun format

       Convert a directory containing raw voice data files, in place,  to  Sun
       format (adds a file header to each file):

       example% audioconvert -p -i voice -f sun *.au

       See attributes(5) for descriptions of the following attributes:

       tab()     allbox;     cw(2.750000i)|    cw(2.750000i)    lw(2.750000i)|
       lw(2.750000i).  ATTRIBUTE TYPEATTRIBUTE  VALUE  ArchitectureSPARC,  x86
       AvailabilitySUNWauda Interface StabilityEvolving

       audioplay(1), audiorecord(1), file(1), attributes(5), largefile(5)

       The  algorithm used for converting multi-channel data to mono is imple-
       mented by simply summing the channels together. If the  input  data  is
       perfectly in phase (as would be the case if a mono file is converted to
       stereo and back to mono), the resulting data may contain  some  distor-

SunOS 5.10                        16 Feb 2001                  audioconvert(1)