Manual page for SNDCONVERT(1)
sndconvert - convert a soundfile to a different format
SYNOPSIS
sndconvert [ 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 ALSO
sndplay(1), sndrecord(1), sndinfo(1), sndcompress(1)
Created by unroff & hp-tools.
© somebody (See intro for details). All Rights Reserved.
Last modified 11/5/97
