---
name: "sndfile-convert"
section: "1"
platform: "linux"
package: "libsndfile"
version: "1.2.2-4"
release: "Arch Linux 2026-11-01"
platforms: ["linux"]
url: "https://new.carta.tech/sndfile-convert"
---

# sndfile-convert(1)

convert sound files from one format to another

## NAME

**sndfile-convert** – convert sound files from one format to another

## SYNOPSIS

**sndfile-convert** \[**-override-sample-rate**=*rate*\] \[**-endian**=**little** \| **big** \| **cpu**\] \[**-normalize**\] \[*encoding*\] *input* *output*

## DESCRIPTION

**sndfile-convert** converts sound files from one audio format to another. The output file is overwritten it it already exists.

### Formats

The format of the output file is determined by the filename extension. The following file formats are currently recognized:

- wav

  WAV (Microsoft)

- aif

  AIFF (Apple/SGI)

- au

  AU (Sun/NeXT)

- snd

  AU (Sun/NeXT)

- raw

  RAW (header-less)

- gsm

  RAW (header-less)

- vox

  RAW (header-less)

- paf

  PAF (Ensoniq PARIS, big-endian)

- fap

  PAF (Ensoniq PARIS, little-endian)

- svx

  IFF (Amiga IFF/SVX8/SV16)

- nist

  SPHERE (NIST SPeech HEader Resources)

- sph

  SPHERE (NIST SPeech HEader Resources)

- voc

  VOC (Creative Labs)

- ircam

  SF (Berkeley/IRCAM/CARL)

- sf

  SF (Berkeley/IRCAM/CARL)

- w64

  W64 (SoundFoundry WAVE 64)

- mat

  MAT4 (GNU Octave 2.0 / Matlab 4.2)

- mat4

  MAT4 (GNU Octave 2.0 / Matlab 4.2)

- mat5

  MAT5 (GNU Octave 2.1 / Matlab 5.0)

- pvf

  PVF (Portable Voice Format)

- xi

  XI (FastTracker 2)

- htk

  HTK (HMM Tool Kit)

- sds

  SDS (Midi Sample Dump Standard)

- avr

  AVR (Audio Visual Research)

- wavex

  WAVEX (MS WAVE with WAVEFORMATEX)

- sd2

  SD2 (Sound Designer II)

- flac

  FLAC (FLAC Lossless Audio Codec)

- caf

  CAF (Apple Core Audio File)

- wve

  WVE (Psion Series 3)

- prc

  WVE (Psion Series 3)

- ogg

  OGG (OGG Container format)

- oga

  OGG (OGG Container format)

- mpc

  MPC (Akai MPC 2k)

- rf64

  RF64 (RIFF 64)

### Options

The following options are recoginzed:

- **-override-sample-rate**=*rate*

  Make the input use sample rate of *rate* Hz.

- **-endian**=**little**

  Make the output file use little endian data.

- **-endian**=**big**

  Make the output file use big endian data.

- **-endian**=**cpu**

  Make the output file use CPU endianness.

- **-normalize**

  Normalize the audio data in the output file.

### Encodings

The optional *encoding* parameter allows setting of the data encoding for the output file. The following encodings are currently supported:

- **-pcms8**

  signed 8 bit pcm

- **-pcmu8**

  unsigned 8 bit pcm

- **-pcm16**

  16 bit pcm

- **-pcm24**

  24 bit pcm

- **-pcm32**

  32 bit pcm

- **-float32**

  32 bit floating point

- **-ulaw**

  ULAW

- **-alaw**

  ALAW

- **-ima-adpcm**

  IMA ADPCM (WAV only)

- **-ms-adpcm**

  MS ADPCM (WAV only)

- **-gsm610**

  GSM6.10 (WAV only)

- **-dwvw12**

  12 bit DWVW (AIFF only)

- **-dwvw16**

  16 bit DWVW (AIFF only)

- **-dwvw24**

  24 bit DWVW (AIFF only)

- **-vorbis**

  Vorbis (OGG only)

If no encoding is specified for the output file, **sndfile-convert** will try to use the encoding of the input file. This will not always work as most container formats (e.g. WAV, AIFF etc) only support a small subset of encodings (e.g. 16 bit PCM, a-law, Vorbis etc).

## EXIT STATUS

The **sndfile-convert** utility exits 0 on success, and \>0 if an error occurs.

## SEE ALSO

**http://libsndfile.github.io/libsndfile/**

## AUTHORS

Erik de Castro Lopo ⟨*erikd@mega-nerd.com*⟩.