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OpenSolaris 2009.06 - man page for audioconvert (opensolaris section 1)

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 for-
       mats. 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  for-
       mat.   If the audio data does not contain a recognizable header, the format must be speci-
       fied 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.

       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

       -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

       -o outfile    Output  File: All input files are concatenated, converted to the output for-
		     mat, 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  pre-
		     ceded  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  spe-
		     cial 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 key-

       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  deter-
		   mine 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
		   linear16,	the  number  of bits of precision. linear16 is typically used for
		   linear32	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 primar-
				ily used for compressing -law voice data (achieving  a	2:1  com-
				pression ratio).

		   g723 	CCITT G.723 compression format. This encoding uses Adaptive Delta
				Pulse Code Modulation (ADPCM) with 3-bit precision. It is primar-
				ily  used  for compressing -law voice data (achieving an 8:3 com-
				pression 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

       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 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

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

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

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

       The algorithm used for converting multi-channel data to mono is implemented by simply sum-
       ming 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.

SunOS 5.11				   16 Feb 2001				  audioconvert(1)

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