afconvert: 16 Encoders, No MP3, and It Drops Your Tags

The most-viewed afconvert question on Stack Overflow, at 34,380 views, asks how to convert a .caf file to MP3. The short answer has not changed since it was asked in January 2009: you can't. What makes it confusing is that afconvert's own format list says you can. On my Mac mini running macOS 26.4.1, afconvert -hf lists 'MPG3' = MPEG Layer 3 (.mp3) with data format '.mp3', and asking for exactly that fails with Error: ExtAudioFileSetProperty ('cfmt') failed ('fmt?').

So I stopped trusting the list and tried every entry in it. -hf prints 24 file formats and 246 file/data format pairs. I converted the same 9.5-second stereo WAV into each one, then asked Core Audio directly which encoders it has. Below are the results, the flags that work, the ones that silently cost you something (tags, mostly), and a batch script for the most common job, FLAC to Apple Lossless.

What afconvert is

/usr/bin/afconvert is Apple's command-line front end to Core Audio's file and codec layer. It ships with every Mac. The binary here is 361,824 bytes and calls itself "Audio File Convert, Version: 2.0, Copyright 2003-2013". Its man page is older and nearly empty: dated February 13, 2007, it documents a single option, -h. Everything useful is in afconvert -h and afconvert -hf, and Apple's longest written explanation of the format list is still Technical Q&A QA1534 from 2009.

The two options you always need are -f (file/container format, a four-character code like m4af) and -d (data format inside it, like aac or alac):

afconvert -f m4af -d aac  in.wav out.m4a      # AAC, default bitrate
afconvert -f m4af -d alac in.flac out.m4a     # FLAC to Apple Lossless
afconvert -f flac -d flac in.wav out.flac     # yes, it writes FLAC
afconvert -f WAVE -d LEI16 in.m4a out.wav     # anything readable to 16-bit WAV
afconvert -f WAVE -d LEI16@44100 in.flac out.wav   # resample on the way

The format list is not an encoder list

Of the 246 pairs, 15 can't be typed: their codes contain NUL bytes ('ms\x00\x02' and friends, the Microsoft ADPCM family), so they can't go on a command line. Of the 231 I could run, 145 wrote a playable file and 86 failed. Almost all failures were the same ('fmt?') error, which says only that the format was rejected.

The reason shows up when you skip afconvert and ask Core Audio. A short Swift program reading kAudioFormatProperty_EncodeFormatIDs and kAudioFormatProperty_DecodeFormatIDs returned 51 decoders and 16 encoders on this Mac:

ENCODE aac ,aace,aacf,aacg,aach,aacl,aacp,alac,alaw,apac,flac,ilbc,ima4,lpcm,opus,ulaw

The -hf list describes what each container can hold, which is mostly what Core Audio can read. Every pair that failed in my run used a codec missing from that encoder line, with two exceptions. iLBC is in the encoder list but only accepts 8 or 16 kHz mono input: my 48 kHz stereo source failed, and an 8 kHz mono copy encoded fine. The other is Ogg, below.

Codec	In -hf list	Can afconvert write it?
MP3, MP2, MP1	Yes (MP3 in 4 containers)	No: no encoder. Decode only
AC-3, E-AC-3	Yes	No: decode only
Vorbis	Yes (CAF, Ogg)	No: decode only
AMR (samr, sawb)	Yes	No: failed at 8 and 16 kHz mono too
AAC (7 variants incl. HE-AAC v1/v2, ELD)	Yes	Yes, in m4a, m4b, mp4, caf, 3gp, 3g2, adts, loas
ALAC	Yes	Yes, in m4a, mp4, caf
FLAC	Yes	Yes, in .flac, m4a, mp4, caf. Not in Ogg
Opus	Yes	Yes in caf and mp4. Not in Ogg
APAC (Apple Positional Audio)	Yes	Yes, in caf and mp4
PCM, u-law, A-law, IMA4	Yes	Yes, in WAV, AIFF, AIFC, CAF, MP4, .snd, .sd2, W64, RF64, BW64 (IMA4 in AIFC and CAF only)

Ogg is its own failure. Core Audio has Opus and FLAC encoders, and -f Oggf accepts them, but every write died with ExtAudioFileWrite failed ('pck?') at every sample rate I tried. Opus in a .caf or .mp4 works. If you need a .opus or .ogg file for something like a Telegram voice note, that still means ffmpeg or opusenc. Reading Ogg is newer: a Hermes Agent pull request from October 3 notes that macOS 15 added an Ogg reader, so afconvert can now decode the files it can't write. I couldn't confirm that one here because I had no way to make an Ogg file to feed it.

Three ways it fails without telling you much

-f without -d fails. afconvert -f WAVE in.wav out.wav and afconvert -f m4af in.wav out.m4a both stop with ExtAudioFileCreateWithURL failed ('fmt?'). The reverse works: -d aac with no -f picks the container from the .m4a extension. This exact bug broke .m4a export in the CCRMA miniAudicle editor; the fix in pull request 88 just adds -d aac.

Some failures leave a 0-byte file. Three pairs (MP2 in AIFC, E-AC-3 in m4a and mp4) failed after creating the output file, leaving an empty file with the name you asked for. Exit code was 1, but a script that checks "does the output exist" will think it worked. Delete the output on non-zero exit.

An .mp3 name doesn't make an MP3. One answer under the 2009 question suggests afconvert -f mp4f -d aac in.caf out.mp3. It exits 0. The result is an MPEG-4 file with the wrong extension; file calls it "ISO Media, MP4 v2" and Apple's own afinfo refuses to open it: Fail: AudioFileOpenURL failed. The answer has a score of -6, deservedly.

It drops your tags

I tagged a FLAC file with title, artist, album and track number using mutagen, then converted it five ways: ALAC m4a, AAC m4a, CAF, WAV and FLAC again. None of the four tags survived in any output. The m4a files gained one tag, iTunSMPB, which is the gapless-playback timing afconvert writes itself. The re-encoded FLAC gained a WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x3 comment and nothing else. The accepted answer on a 2013 Stack Overflow question about keeping tags says afconvert "loses all tags", and that's still true on macOS 26. Plan to re-tag after converting a music library.

Sizes and speed on a 10-minute file

For timing I stretched the source into a 606.9-second, 48 kHz 16-bit stereo WAV (116.5 MB) and encoded it each way with /usr/bin/time. The audio is synthesized speech from the say command, not music, so treat the compression ratios as a lower bound on what music would take.

Everything ran at 200 to 700 times real time. Default AAC came out at about 101 kbps for 48 kHz stereo, lower than most people expect, so pass -b or use VBR. The VBR line I used was -s 3 -q 127 -u vbrq 127, which landed at 140 kbps. ALAC was 9% larger than FLAC on this file, which matches the complaint in an Ask Different thread about ALAC files being "too large": that's the codec, not a wrong flag.

Lossless means lossless here. I decoded the FLAC and ALAC files back to WAV and hashed the PCM: both matched the source exactly (MD5 24f7ad9b…, 29,129,088 frames each). The AAC decode had the same frame count, so afconvert trims encoder priming on the way back. Two details on WAV output: afconvert writes a WAVE_FORMAT_EXTENSIBLE header plus a 4,020-byte FLLR padding chunk, so its WAV files are 4,052 bytes bigger than a minimal one. --no-filler removes the padding; the extensible header stays. Most software doesn't care. A naive WAV parser that assumes a 44-byte header will.

Batch: FLAC to ALAC for a folder

The second most-viewed afconvert question (28,387 views) is about converting a whole directory (WAV to CAF, in 2009); a Super User thread asks the same for FLAC. This script writes a .m4a next to every .flac under a folder, skips files already done, and removes the 0-byte leftovers described above. I ran it twice on a test tree with spaces in folder and file names: the first run converted 2, the second skipped both.

#!/bin/zsh
# flac2alac.zsh <dir>: write an ALAC .m4a next to every .flac under <dir>.
# afconvert drops tags, so re-tag afterwards with a tagger of your choice.
setopt extended_glob null_glob
ok=0 fail=0
for f in "$1"/**/*.flac; do
  out="${f%.flac}.m4a"
  if [[ -e "$out" ]]; then echo "skip  $out"; continue; fi
  if afconvert -f m4af -d alac "$f" "$out" 2>/tmp/flac2alac.err; then
    ok=$((ok+1))
  else
    rm -f "$out"   # a failed run can leave a 0-byte file behind
    echo "FAIL  $f: $(cat /tmp/flac2alac.err)"; fail=$((fail+1))
  fi
done
echo "converted $ok, failed $fail"

One flag I expected to help didn't: -d N ("native format of the lossless encoded source") only works when the output is PCM. -f flac -d N on an ALAC file failed with ('fmt?'), while -f WAVE -d N on a FLAC file worked.

When to use ffmpeg instead

afconvert is the right tool when the target is AAC, ALAC, FLAC, WAV or AIFF and you don't care about tags. It's already installed, it's fast, and its AAC encoder is Apple's. For MP3, Ogg Vorbis, .opus files, AC-3, or anything where tags must survive, you need ffmpeg (brew install ffmpeg) or LAME. That's the same answer the top-voted Stack Overflow reply gave in 2009, quoting an Apple Core Audio mailing-list post that "Apple does not ship any form of MP3 encoder." It's the same pattern I found with sips and WebP: a macOS tool that reads far more formats than it writes, and doesn't say which is which until you try. I didn't benchmark ffmpeg against afconvert for this post because ffmpeg isn't installed on this machine.

For a sense of what people actually trip on: the Stack Exchange API title search for "afconvert" on Stack Overflow, Super User, Ask Different and Unix & Linux returns 20 questions with 127,629 views in total. The MP3 question alone is 27% of those views. Others cover AC-3 output (neither answer uses afconvert, because it can't), recursive FLAC encoding, and which AAC code means what.

If you're scripting other built-in tools, textutil and hdiutil have their own surprises worth knowing before you script them.

FAQ

Can afconvert convert to MP3?

No. afconvert can read MP3 but macOS has no MP3 encoder, so -f MPG3 -d .mp3 fails with ExtAudioFileSetProperty ('cfmt') failed ('fmt?') even though afconvert -hf lists MP3. On macOS 26.4.1, Core Audio reports 16 encoders and none is MP3. Use ffmpeg or LAME for MP3, or convert to AAC in an .m4a file with afconvert -f m4af -d aac.

How do I convert FLAC to ALAC with afconvert?

Run afconvert -f m4af -d alac input.flac output.m4a. The audio is bit-identical to the FLAC source when decoded, but afconvert does not copy tags such as title, artist or album, so re-tag the .m4a afterwards. Expect the ALAC file to be a little larger than the FLAC; in my test it was 9% larger.

Does afconvert keep metadata?

No. In tests on macOS 26.4.1, title, artist, album and track number tags from a FLAC file were dropped when converting to ALAC, AAC, CAF, WAV and FLAC. afconvert only adds its own technical tags, such as iTunSMPB gapless information in .m4a files. Use ffmpeg or a tag editor if metadata must carry over.

Method: all tests ran on 2026-10-09 between 12:00 and 12:10 KST on a Mac mini M4 (Mac16,10, macOS 26.4.1 build 25E253, /usr/bin/afconvert 361,824 bytes, man page dated February 13, 2007) in a scratch folder under my work directory. The source was a 9.48-second clip from say, converted to 48 kHz 16-bit stereo; the format matrix fed it to every pair printed by afconvert -hf and recorded exit code, stderr and output size. Failures for AMR, iLBC, MP3 and Ogg were retried at 8 kHz mono, 16 kHz mono and 44.1 kHz stereo. The encoder and decoder lists come from AudioFormatGetProperty in a small Swift program. Timings are single wall-clock runs, and file sizes are for speech, not music. Tag tests used mutagen 1.48.1 to write and read Vorbis comments and MP4 atoms. The Stack Exchange counts come from the API title search on the same day. I did not test Ogg decoding, multichannel files, or ffmpeg.