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  standard input stream and writes an audio file to
     the standard output. Otherwise, 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 recognizable header, the format must be  specified
     with  the  -i option, using the rate, encoding, and channels
     keywords to identify the input data format.

     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  resulting  format  and con-
     catenated together. The output file will  contain  an  audio
     file  header,  unless  format=raw is specified in the output
     format options.

     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 out-
     put 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 rewrit-
           ten. If a target file is a symbolic link, the underly-
           ing  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 specifica-
           tion 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 specifi-
           cation, until a new input format is specified.

     file  File Specification: The named  audio  files  are  con-
           catenated, 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 input.

     -?    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.

           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.

           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 telephone quality speech.

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



           Linear Pulse Code Modulation (PCM) encoding. The  name
           identifies  the  number of bits of precision. linear16
           is typically used for high quality audio data.

     pcm   Same as linear16.

     g721  CCITT G.721 compression  format.  This  encoding  uses
           Adaptive  Delta  Pulse Code Modulation (ADPCM) with 4-
           bit precision. It is primarily  used  for  compressing
           -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 precision. It is primarily  used  for  compressing
           -law  voice data (achieving an 8:3 compression ratio).
           The audio quality 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 encoding:

     voice Equivalent to encoding=ulaw,rate=8k,channels=mono.

     cd    Equivalent                                          to

     dat   Equivalent                                          to

           This option specifies the  audio  file  format.  Valid
           formats 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.

           (-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 bytes).


     Example 1: Recording and compressing voice data before stor-
     ing it

     Record voice data and compress it before  storing  it  to  a

     example% audiorecord | audioconvert -f g721 >

     Example 2: Concatenating two audio files

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

     example% audioconvert -f ulaw,rate=16k,mono -o 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  attri-

    |       ATTRIBUTE TYPE        |       ATTRIBUTE VALUE       |
    | Architecture                | SPARC, x86                  |
    | Availability                | SUNWauda                    |
    | Interface Stability         | Evolving                    |


     audioplay(1), audiorecord(1), file(1), attributes(5), large-


     The algorithm used for converting multi-channel data to mono
     is  implemented  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 distortion.

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