Name

rdconvert — Convert an audio file to a different format

Synopsis

rdconvert [OPTIONS] {src-file}

Description

rdconvert(1) can be used to convert audio files between different formats.

Options

--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.

See Also

rdexport(1) , rdimport(1) , rdmarkerset(8)