/Applications directory - and run the afconvert
command. The afconvert man page
provides minimal information on the program's usage; issue the command
afconvert -help at a Terminal window shell prompt to obtain
details on the program's usage.$ afconvert -help
Audio File Convert
Version: 2.0
Copyright 2003-2013, Apple Inc. All Rights Reserved.
Specify -h (-help) for command options
Usage:
afconvert [option...] input_file [output_file]
Options may appear before or after the direct arguments. If output_file
is not specified, a name is generated programmatically and the file
is written into the same directory as input_file.
afconvert input_file [-o output_file [option...]]...
Output file options apply to the previous output_file. Other options
may appear anywhere.
General options:
{ -d | --data } data_format[@sample_rate][/format_flags][#frames_per_packet]
[-][BE|LE]{F|[U]I}{8|16|24|32|64} (PCM)
e.g. BEI16 F32@44100
or a data format appropriate to file format (see -hf)
format_flags: hex digits, e.g. '80'
Frames per packet can be specified for some encoders, e.g.: samr#12
A format of "0" specifies the same format as the source file,
with packets copied exactly.
A format of "N" specifies the destination format should be the
native format of the lossless encoded source file (alac, FLAC only)
{ -c | --channels } number_of_channels
add/remove channels without regard to order
{ -l | --channellayout } layout_tag
layout_tag: name of a constant from CoreAudioTypes.h
(prefix "kAudioChannelLayoutTag_" may be omitted)
if specified once, applies to output file; if twice, the first
applies to the input file, the second to the output file
{ -b | --bitrate } total_bit_rate_bps
e.g. 256000 will give you roughly:
for stereo source: 128000 bits per channel
for 5.1 source: 51000 bits per channel
(the .1 channel consumes few bits and can be discounted in the
total bit rate calculation)
{ -q | --quality } codec_quality
codec_quality: 0-127
{ -r | --src-quality } src_quality
src_quality (sample rate converter quality): 0-127 (default is 127)
{ --src-complexity } src_complexity
src_complexity (sample rate converter complexity): line, norm, bats
{ -s | --strategy } strategy
bitrate allocation strategy for encoding an audio track
0 for CBR, 1 for ABR, 2 for VBR_constrained, 3 for VBR
--prime-method method
decode priming method (see AudioConverter.h)
--prime-override samples_prime samples_remain
can be used to override the priming information stored in the source
file to the specified values. If -1 is specified for either, the value
in the file is used.
--no-filler
don't page-align audio data in the output file
--soundcheck-generate
analyze audio, add SoundCheck data to the output file
--media-kind "media kind string"
media kinds are: "Audio Ad", "Video Ad"
--anchor-loudness
set a single precision floating point value to
indicate the anchor loudness of the content in dB
--generate-hash
generate an SHA-1 hash of the input audio data and add it to the output file.
--codec-manuf codec_manuf
specify the codec with the specified 4-character component manufacturer
code
--dither algorithm
algorithm: 1-2
--mix
enable channel downmixing
{ -u | --userproperty } property value
set an arbitrary AudioConverter property to a given value
property is a four-character code; value can be a signed
32-bit integer or a single precision floating point value.
e.g. '-u vbrq <sound_quality>' sets the sound quality level
(<sound_quality>: 0-127)
May not be used in a transcoding situation.
-ud property value
identical to -u except only applies to a decoder. Fails if there is no
decoder.
-ue property value
identical to -u except only applies to an encoder. Fails if there is no
encoder.
Input file options:
--read-track track_index
For input files containing multiple tracks, the index (0..n-1)
of the track to read and convert.
--offset number_of_frames
the starting offset in the input file in frames. (The first frame is
frame zero.)
--soundcheck-read
read SoundCheck data from source file and set it on any destination
file(s) of appropriate filetype (.m4a, .caf).
--copy-hash
copy an SHA-1 hash chunk, if present, from the source file to the output file.
--gapless-before filename
file coming before the current input file of a gapless album
--gapless-after filename
file coming after the current input file of a gapless album
Output file options:
-o filename
specify an (additional) output file.
{ -f | --file } file_format
use -hf for a complete list of supported file/data formats
--condensed-framing field_size_in_bits
specify storage size in bits for externally framed packet sizes.
Supported value is 16 for aac in m4a file format.
Other options:
{ -v | --verbose }
print progress verbosely
{ -t | --tag }
If encoding to CAF, store the source file's format and name in a user
chunk. If decoding from CAF, use the destination format and filename
found in a user chunk.
{ --leaks }
run leaks at the end of the conversion
{ --profile }
collect and print performance information
Help options:
{ -hf | --help-formats }
print a list of supported file/data formats
{ -h | --help }
print this help
$If you want to see the file formats and audio encodings it supports, you can
issue the command afconvert -hf or afconvert --help-formats
.
$ afconvert -hf
Audio file and data formats:
'3gpp' = 3GP Audio (.3gp)
data_formats: 'Qclp' 'aac ' 'aace' 'aacf' 'aach' 'aacl'
'aacp' 'samr'
'3gp2' = 3GPP-2 Audio (.3g2)
data_formats: 'Qclp' 'aac ' 'aace' 'aacf' 'aach' 'aacl'
'aacp' 'samr'
'adts' = AAC ADTS (.aac, .adts)
data_formats: 'aac ' 'aach' 'aacp'
'ac-3' = AC3 (.ac3)
data_formats: 'ac-3'
'AIFC' = AIFC (.aifc, .aiff, .aif)
data_formats: I8 BEI16 BEI24 BEI32 BEF32 BEF64 UI8 'ulaw'
'alaw' 'MAC3' 'MAC6' 'ima4' 'QDMC' 'QDM2'
'Qclp' 'agsm'
'AIFF' = AIFF (.aiff, .aif)
data_formats: I8 BEI16 BEI24 BEI32
'amrf' = AMR (.amr)
data_formats: 'samr' 'sawb'
'm4af' = Apple MPEG-4 Audio (.m4a, .m4r)
data_formats: 'aac ' 'aace' 'aacf' 'aach' 'aacl' 'aacp'
'alac' 'paac'
'm4bf' = Apple MPEG-4 AudioBooks (.m4b)
data_formats: 'aac ' 'aace' 'aacf' 'aach' 'aacl' 'aacp'
'paac'
'caff' = CAF (.caf)
data_formats: '.mp1' '.mp2' '.mp3' 'QDM2' 'QDMC' 'Qclp'
'Qclq' 'TS\x00\x02' 'TS\x00\x11' 'TS\x00E'
'WMA1' 'WMA2' 'WMA3' 'WMAL' 'aac ' 'aace'
'aacf' 'aach' 'aacl' 'aacp' 'alac' 'alaw'
'dvi8' 'ilbc' 'ima4' I8 BEI16 BEI24 BEI32
BEF32 BEF64 LEI16 LEI24 LEI32 LEF32 LEF64
'ms\x00\x02' 'ms\x00\x11' 'ms\x001' 'paac'
'qaac' 'qach' 'qacp' 'samr' 'ulaw' 'zaac'
'zach' 'zacp'
'MPG1' = MPEG Layer 1 (.mp1, .mpeg, .mpa)
data_formats: '.mp1'
'MPG2' = MPEG Layer 2 (.mp2, .mpeg, .mpa)
data_formats: '.mp2'
'MPG3' = MPEG Layer 3 (.mp3, .mpeg, .mpa)
data_formats: '.mp3'
'mp4f' = MPEG-4 Audio (.mp4)
data_formats: 'aac ' 'aace' 'aacf' 'aach' 'aacl' 'aacp'
'NeXT' = NeXT/Sun (.snd, .au)
data_formats: I8 BEI16 BEI24 BEI32 BEF32 BEF64 'ulaw'
'Sd2f' = Sound Designer II (.sd2)
data_formats: I8 BEI16 BEI24 BEI32
'WAVE' = WAVE (.wav)
data_formats: UI8 LEI16 LEI24 LEI32 LEF32 LEF64 'ulaw'
'alaw'
$For the supported file formats, you will see the name, e.g. "WAVE", the
recognized file extensions, e.g., .wav for
Waveform Audio
File Format, i.e. WAVE or WAV, which reflects the commonly used file
extension, then the data formats, which are UI8, LEI16 LEI24 LEI32 LEF32 LEF64
'ulaw', and 'alaw' for WAV files. The data formats are character codes such as
"ulaw" or all capital letter strings for a
Linear pulse-code modulation (LPCM) format. The LPCM formats are
descibed in terms of endianness, value type, and sample size in bits. On PCs,
LPCM often refers to the format used in WAV and
AIFF audio
container formats, but LPCM data may also be stored in other formats such as
AU,
raw audio format, and various multimedia container formats. The
μ-law encoding is a very old encoding which reduces the dynamic
range of an audio signal. The
A-law algorithm
is also used to modify the
dynamic range of an
analog signal. For further information on the afconvert utility, see
pages 97 through 99 of
Learning Core Audio: A Hands-On Guide to Audio Programming for Mac and iOS 1st Edition
If I have an
MPEG-4 Part 14 .m4a file that I would like to convert to a WAV file,
I can use the afconvert command below. The utility recognizes I want a
WAV file from the extension and the data format for the output file is
specified with -d LEI16. If I then wish to view information
on the example.wav output file, I can examine it with the
afinfo command - see
Obtaining info on an audio file
with afinfo.
$ afconvert "example.m4a" -d LEI16 "example.wav"
$ afinfo example.wav
File: example.wav
File type ID: WAVE
Num Tracks: 1
----
Data format: 2 ch, 44100 Hz, 'lpcm' (0x0000000C) 16-bit little-endian signed integer
no channel layout.
estimated duration: 4.085261 sec
audio bytes: 720640
audio packets: 180160
bit rate: 1411200 bits per second
packet size upper bound: 4
maximum packet size: 4
audio data file offset: 4096
optimized
source bit depth: I16
----
$The afinfo command shows the duration of the audio file, which in this
case is merely a few seconds, and the bit rate. The data rate, which is
"16-bit little-endian signed integer" and the source bit depth, "I16" are also
displayed. They were specified with the -d LEI16 parameter to the
afconvert command; the "LE" stands for
"Little Endian" and the "I" stands for
"Integer". The codes for the LPCM formats, such as the UI8 LEI16 LEI24 LEI32
LEF32 LEF64 formats for WAV files, can be decoded from the following table:
| LE | Little Endian |
| BE | Big Endian |
| F | Floating point |
| I | Integer |
| UI | Unsigned Integer |
| 8/16/24/32/64 | Number of bits |
If I wanted to listen to the audio file using a command line utility, I could use the afplay command to play it - see Audio File Play - afplay.
Suppose I wanted to convert the .m4a file to an
MP3 file,
instead. From the output of the afconvert -hf command it appears
the MP3 format is supported. But, if I try to convert the file to an .mp3 file,
I get an error message:
$ afconvert "example.m4a" -d .mp3 "example.mp3"
Error: ExtAudioFileSetProperty ('cfmt') failed ('fmt?')
$So why did I get an error message when the MP3 format appears to be supported? Some formats have restrictions. From Apple's Core Audio Services web page:
In many cases, you use Extended Audio File Services, which provides the simplest interface for reading and writing audio data. Files read using this API are automatically uncompressed and/or converted into linear PCM format, which is the native format for audio units. Similarly, you can use one function call to write linear PCM audio data to a file in a compressed or converted format. Supported Audio File and Data Formats in OS X lists the file formats that Core Audio supports by default. Some formats have restrictions; for example, by default, Core Audio can read, but not write, MP3 files.
You can see audio data types for which the utility can not encode from LCPM in the following table from Supported Audio File and Data Formats in OS X:
Audio data type |
Encode from linear PCM? | Decode to linear PCM? |
|---|---|---|
MPEG Layer 3 ( |
No | Yes |
MACE 3:1 ( |
Yes | Yes |
MACE 6:1 ( |
Yes | Yes |
QDesign Music 2 ( |
Yes | Yes |
QDesign ( |
No | Yes |
Qualcomm PureVoice ( |
Yes | Yes |
Qualcomm QCELP ( |
No | Yes |
AAC ( |
Yes | Yes |
Apple Lossless ( |
Yes | Yes |
Apple GSM 10:1 ( |
No | Yes |
ALaw 2:1 ( |
Yes | Yes |
Apple DRM Audio Decoder ( |
No | Yes |
AC-3 |
No | No |
DVI 4:1 ( |
No | Yes |
Apple IMA 4:1 ( |
Yes | Yes |
LPC 23:1 ( |
No | Yes |
Microsoft ADPCM |
No | Yes |
DVI ADPCM |
Yes | Yes |
GSM610 |
No | Yes |
AMR Narrowband ( |
Yes | Yes |
μLaw 2:1 ( |
Yes | Yes |