SNDCONVERT(1)SNDCONVERT(1)NAMEsndconvert - convert a soundfile to a different format
SYNOPSISsndconvert [ option ] ... file
DESCRIPTION
Sndconvert modifies the file according to various options. By default,
a consistency check is performed; the various options allow conversion
between formats, including non-soundfile formats.
OPTIONS-o outfile
This option indicates that the output is to be written to the
specified file. By default, soundfiles are modified in place,
with the exception of a conversion that changes the file type
(i.e. -d, -m, -r, and -a options).
-i infostring
This option specifies that its argument is to be appended to the
info string in the soundfile header. By default, the info string
is null.
-f formatCode
This specifies the resulting data format of the sound. The value
of the number represents the format code; format codes are
defined in <sound/soundstruct.h>. The default format code (3)
corresponds to 16 bit linear data. Some useful conversions are
SND_FORMAT_LINEAR_16 (3) to SND_FORMAT_MULAW_8 (1)
SND_FORMAT_MULAW_8 (1) to SND_FORMAT_LINEAR_16 (3)
SND_FORMAT_FLOAT (6) to SND_FORMAT_LINEAR_16 (3)
Format conversions are carried out by the sound library function
SNDConvertSound(). See the Reference manual and the Sound
release notes for further specifications.
-g scaleExponent
This allows the user to determine the scaling on the float-to-
integer or integer-to-float translations. The scale factor used
in translating float-to-integer is (float) * ((1 <<
scaleExponent)-1). The scale factor used in translating
integer-to-float is (integer) / ((1 << scaleExponent)-1). The
default scaleExponent is 15, which means that a float value of
1.0 is translated to the integer value of 32767. Using a
scaleExponent of 23 translates a float value of 1.0 to a full-
scale integer value used by the DSP56001.
-s sampleRate
This specifies the resulting sampling rate of the sound. It may
be a floating point number. If the source file sampling rate is
not equal to sampleRate, then sampling rate conversion is
performed.
-c channelCount
This specifies the resulting number of channels of the sound.
The default channel count is 2. If the input file is mono (1
channel) and the channel count is 2, then a mono-to-stereo
conversion is performed, for linear 16 bit files.
-r This specifies the input file to be a raw data file, to which a
header is added. The resulting soundfile is of the same name as
the input file with a .snd extension.
-a This option specifies the input file to be a raw data file, as
with the -r option, but the data format is ascii. Each sound
sample is specified as a number in the range -1.0 to 1.0, and
the numbers are separated by newlines. The output files that
can be generated from ascii input files are SND_FORMAT_LINEAR_16
(3), SND_FORMAT_MULAW_8 (1), SND_FORMAT_FLOAT (6).
-d This option indicates that the file is a DSP absolute load
image. The file must have a .lod extension, and the output file
will be of the same name with a .snd extension.
FILES
The <sound/soundstruct.h> file contains the data format codes defined
by the system.
SEE ALSOsndplay(1), sndrecord(1), sndinfo(1), sndcompress(1)NeXT Computer, Inc. August 13, 1990 SNDCONVERT(1)
