rdconvert — Convert an audio file to a different format
--destination-bit-rate=bit-rate
Use a bit rate of bit-rate bits per
second. This option is ignored for PCM and FLAC formats, and is
mutually exclusive with the --destination-quality
option. The default value is 0.
--destination-channels=chans
Use chans channels. Supported values
are 1 and 2. The
default value is 2.
--destination-file=filename
Write the converted data to filename.
If not specified, the data will be written to the name of the
input file with the default extension of the destination format
appended.
--destination-format=format
Write the converted data to the specified format.
format can be one of the following:
0PCM16 WAV
2MPEG Layer 2 (Raw)
3MPEG Layer 3 (Raw)
4Free Lossless Audio Codec (FLAC)
5OggVorbis
6MPEG Layer 2 (BWF WAV Container)
7PCM24 WAV
--destination-quality=qual
Use a variable bitrate with a quality of
chans. Supported values
are -1 through 10.
This parameter is used only with a format of 5
(OggVorbis). The default value is 0.
--destination-sample-rate=rate
Use a sample rate of rate samples per
second. Not all sample rates are supported for all formats; see
the relevant MPEG specifications for details. The default value is
48000.
--end-point=msec
Stop converting the audio data at the point
msec mS from the start of the source
file. A value of -1 means to continue
conversion to the end of the source file, which is the default.
--normalization-level=lvl
Peak-normalize the audio to lvl dBFS.
A value of 0 disables normalization, which
is the default.
--speed-ratio=ratio
Alter the tempo of the audio by ratio.
A value of 1.0 specifies no tempo alteration,
which is the default.
--start-point=msec
Start converting the audio data at the point
msec mS into the source file. The
default value is 0.