audioconvert(1) 						   User Commands						   audioconvert(1)

NAME

       audioconvert - convert audio file formats

SYNOPSIS

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

DESCRIPTION

       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 recogniz-
       able 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 concatenated 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 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.

OPTIONS

       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  unspeci-
		       fied 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, converted to the output format, and written to the named output file. If -o
		       and -p are not specified, the concatenated 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 format describes all input files following that  specification,  until  a
		       new input format is specified.

       file	       File  Specification:  The  named  audio files are concatenated, 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). Standard 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 representation. Encodings determine precision implicitly (ulaw  implies  8-bit
		       precision) or explicitly as part of the name (for example, linear16). Valid encoding values are:

		       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.

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

		       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 encoding=linear16,rate=44.1k,channels=stereo.

		       dat	Equivalent to encoding=linear16,rate=48k,channels=stereo.

       format	       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 conjunction 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.

USAGE

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

EXAMPLES

       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 format, 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

ATTRIBUTES

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

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

SEE ALSO

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

NOTES

       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 dis-
       tortion.

SunOS 5.10							    16 Feb 2001 						   audioconvert(1)