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:
0
PCM16 WAV
2
MPEG Layer 2 (Raw)
3
MPEG Layer 3 (Raw)
4
Free Lossless Audio Codec (FLAC)
5
OggVorbis
6
MPEG Layer 2 (BWF WAV Container)
7
PCM24 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
.