Convert M4A to AIF — Free Online Tool
Convert M4A files (AAC-encoded audio from iTunes or podcasts) to AIF, Apple's uncompressed PCM format, producing a lossless master-quality file stored as 16-bit big-endian PCM. This is ideal when you need to bring compressed iTunes audio into professional audio workstations or mastering environments that require uncompressed source files.
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FFmpeg Command
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Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
M4A files store audio using AAC compression, a lossy codec that discards audio data to reduce file size. Converting to AIF decodes that AAC stream back to raw PCM samples and writes them into the AIF container using 16-bit big-endian PCM (pcm_s16be). This is a full decode-and-re-encode operation, not a remux — the AAC bitstream cannot be directly placed into an AIF container because AIF only supports uncompressed PCM codecs. The resulting file will be significantly larger (typically 8–10x the size of the M4A), and because the source was lossy AAC, the AIF will not recover any audio detail that was discarded during the original AAC encoding. What you get is an uncompressed representation of the decoded AAC audio, suitable for editing in DAWs or workflows that cannot accept compressed formats.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles the decoding of the AAC audio from the M4A container and the encoding of uncompressed PCM into the AIF container. |
-i input.m4a
|
Specifies the input M4A file. FFmpeg reads the MPEG-4 container, identifies the AAC audio stream inside it, and prepares it for decoding. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio format used in AIF files. This tells FFmpeg to fully decode the AAC audio and write raw PCM samples in the byte order expected by Apple's AIF container format. |
output.aif
|
Specifies the output filename and triggers FFmpeg to write an AIF container. The .aif extension causes FFmpeg to use the AIF muxer, which wraps the pcm_s16be audio stream in Apple's Audio Interchange File Format structure. |
Common Use Cases
- Importing iTunes-purchased M4A tracks into a professional DAW like Pro Tools or Logic Pro as uncompressed AIF stems for mixing or mastering sessions
- Converting podcast M4A source files to AIF so they can be loaded into broadcast audio editors that require uncompressed input, such as Adobe Audition in certain hardware-triggered workflows
- Preparing M4A audiobook or soundtrack files for audio restoration software that only accepts PCM-based formats like AIF or WAV
- Archiving a decoded version of AAC-compressed music to AIF before applying non-destructive edits in a waveform editor, ensuring no further generation loss during the editing process
- Converting M4A recordings from an iPhone voice memo (which exports as AAC in an M4A container) to AIF for use in a Mac-based audio post-production pipeline that requires native Apple uncompressed format
- Delivering audio assets to a sound designer or studio that has a strict policy of accepting only uncompressed AIF files, even when the original source material was AAC-compressed
Frequently Asked Questions
No. The M4A file stores audio using AAC, which is a lossy codec — it permanently discards audio information during encoding to achieve smaller file sizes. Converting to AIF decodes the AAC audio back to uncompressed PCM, but it cannot restore the data that was already discarded. The AIF file will be an exact uncompressed representation of the decoded AAC audio, not a higher-quality version of the original recording. You will get a much larger file with the same perceptible quality as the source M4A.
M4A files use AAC compression, which typically achieves compression ratios of 8:1 to 12:1 compared to uncompressed audio. AIF stores audio as raw uncompressed 16-bit PCM samples at the full sample rate, with no compression applied. A typical 128kbps AAC M4A file will expand to roughly 1411kbps as a 16-bit 44.1kHz AIF — meaning a 5MB M4A track may become 40–50MB as an AIF. This is expected and is why lossless uncompressed formats are used for editing rather than storage.
Most iTunes metadata tags embedded in the M4A (such as artist, album, title, and track number) will not be reliably preserved in the AIF output. The AIF format has limited metadata support compared to M4A's robust iTunes tagging system, and FFmpeg does not map all M4A metadata atoms to AIF chunks during this conversion. Additionally, M4A chapter markers are not supported by the AIF format and will be dropped. If metadata preservation is critical, consider tagging the AIF file manually after conversion using a tool like Meta or MusicBrainz Picard.
Yes. The default command uses pcm_s16be (16-bit signed big-endian PCM), but you can substitute a higher bit depth codec by changing the -c:a flag. For 24-bit output use -c:a pcm_s24be, for 32-bit integer use -c:a pcm_s32be, or for 32-bit float use -c:a pcm_f32be. For example: ffmpeg -i input.m4a -c:a pcm_s24be output.aif. Keep in mind that since the source is 16-bit AAC audio, increasing the bit depth will not add real audio resolution — it simply stores the decoded samples in a wider container, which some DAWs or workflows may prefer.
On macOS or Linux you can use a shell loop: for f in *.m4a; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m4a}.aif"; done. On Windows Command Prompt use: for %f in (*.m4a) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". Each file is processed sequentially, and the output AIF is named to match the input M4A. Note that this browser-based tool processes one file at a time; the FFmpeg command approach on your desktop is the recommended method for batch jobs.
Both AIF and WAV are uncompressed PCM formats and are functionally equivalent in audio quality at the same bit depth and sample rate. AIF is Apple's native format and has historically had slightly better metadata and loop marker support in macOS and Logic Pro, making it a natural choice for Mac-based workflows. WAV is more universally compatible across Windows and Linux tools. If your work stays entirely within a Mac or Apple ecosystem, AIF is the conventional choice; for cross-platform exchange, WAV is often preferred.
Technical Notes
The AIF format uses big-endian byte ordering, which is why FFmpeg selects pcm_s16be (signed 16-bit big-endian) as the default codec rather than the little-endian pcm_s16le used in WAV files. This byte order is a historic artifact of AIF's origins on Motorola 68k-based Macs. The conversion process involves fully decoding the AAC bitstream from the M4A container — using FFmpeg's built-in AAC decoder — and then writing raw PCM frames into an AIF container. There is no intermediate lossy step, so no additional quality degradation occurs beyond what AAC already introduced. M4A files can technically contain other codecs (FLAC, Opus, Vorbis), but in practice nearly all M4A files use AAC. The -vn flag is not required here since M4A is an audio-only format, but the special_flags field notes it as a precaution for any edge-case M4A files that might contain embedded video thumbnails. The AIF format does not support chapters, gapless playback metadata, or the iTunes-specific tagging atoms present in M4A, so those features are lost in the output. If the source M4A was encoded at 44.1kHz stereo (the iTunes standard), the output AIF will match that sample rate and channel layout automatically without needing additional flags.