Convert AIFF to MP3 — Free Online Tool
Convert AIFF audio files to MP3 directly in your browser using the LAME encoder (libmp3lame), reducing uncompressed PCM audio to a compact, universally compatible lossy format. Ideal for shrinking large Apple lossless audio files for streaming, sharing, or playback on any device.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AIFF file here
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Settings
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
AIFF stores audio as uncompressed PCM data (typically 16-bit or 24-bit big-endian samples), which means the file size is directly proportional to duration and sample depth with no compression applied. During this conversion, FFmpeg decodes the raw PCM stream from the AIFF container and passes it through the LAME MP3 encoder (libmp3lame), which applies perceptual audio compression using psychoacoustic modeling to discard frequencies less audible to the human ear. The result is an MP3 file encoded at 128 kbps by default — roughly 1 MB per minute — compared to an uncompressed AIFF which can run 10 MB per minute or more at 16-bit/44.1kHz stereo. This is a lossy transcode: once converted, the discarded audio data cannot be recovered, so keep your original AIFF if archival quality matters.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser version of this tool, the same FFmpeg engine runs via WebAssembly (FFmpeg.wasm), so the command is identical to what you would run on your local desktop. |
-i input.aiff
|
Specifies the input file — an AIFF file containing uncompressed big-endian PCM audio (typically 16-bit or 24-bit). FFmpeg automatically detects the AIFF container and the PCM codec variant used. |
-c:a libmp3lame
|
Selects the LAME MP3 encoder for the audio stream. libmp3lame is the industry-standard open-source MP3 encoder and produces the most compatible MP3 output — required here because AIFF's raw PCM audio must be re-encoded rather than simply remuxed. |
-b:a 128k
|
Sets the MP3 audio bitrate to 128 kilobits per second, the default for this tool. This produces a file roughly 10 times smaller than a 16-bit/44.1kHz stereo AIFF with quality suitable for general music listening. Change this value (e.g., to 192k or 320k) to trade file size for higher audio fidelity. |
output.mp3
|
Defines the output filename and tells FFmpeg to write an MPEG Audio Layer III file. The .mp3 extension causes FFmpeg to use the MP3 container, which wraps the libmp3lame-encoded audio and supports ID3 tags for metadata like title, artist, and album. |
Common Use Cases
- Exporting finished music tracks from a macOS DAW (like Logic Pro, which natively outputs AIFF) into MP3 for distribution on streaming platforms or digital download stores.
- Reducing the file size of a large AIFF field recording or podcast master before uploading to a podcast hosting service that enforces file size limits.
- Converting an AIFF audio sample library to MP3 so it can be used in web-based or mobile apps that do not support AIFF playback.
- Preparing AIFF voiceover or narration recordings for email delivery or embedding in a PowerPoint/Keynote presentation where file size is a concern.
- Converting AIFF music files ripped from CD on a Mac into MP3 so they can be played on older car stereos, MP3 players, or Android devices that lack AIFF support.
- Archiving a large AIFF interview recording to MP3 at a higher bitrate (e.g., 192k) for long-term storage at reduced disk footprint without retaining the full uncompressed master.
Frequently Asked Questions
The size reduction depends on the AIFF's bit depth and the MP3 bitrate you choose. A standard 16-bit/44.1kHz stereo AIFF runs about 10 MB per minute uncompressed. At the default 128 kbps MP3 setting, the same audio becomes roughly 1 MB per minute — approximately a 10x reduction. At 320 kbps (the highest MP3 quality), you still get about a 4x reduction. Higher bit-depth AIFF files (24-bit or 32-bit) will see even greater size savings.
Yes — MP3 is a lossy format and does discard some audio data. For most listening purposes at 128 kbps or higher, the difference is imperceptible on typical speakers or headphones. However, audiophiles or professional engineers doing further processing may notice artifacts, especially in high-frequency content or complex transients. If the audio will be re-edited or re-encoded later, it's best to keep the original AIFF and only export MP3 as a delivery copy.
AIFF stores metadata in AIFF-C chunks, while MP3 uses ID3 tags. FFmpeg will attempt to map compatible metadata fields (such as title, artist, and album) from the AIFF source into ID3 tags in the output MP3. However, not all AIFF metadata fields have direct ID3 equivalents, so some custom or non-standard tags may be lost. You can verify and edit the ID3 tags in the resulting MP3 using a tag editor like Mp3tag or MusicBrainz Picard.
Replace the value after -b:a in the command. For example, to encode at 320 kbps for the highest MP3 quality, use: ffmpeg -i input.aiff -c:a libmp3lame -b:a 320k output.mp3. Common options are 64k (voice/speech), 128k (general listening, the default), 192k (good quality music), and 320k (near-transparent quality). Increasing the bitrate improves quality but also increases file size proportionally.
Yes. On Linux or macOS, you can loop over all AIFF files in a folder with: for f in *.aiff; do ffmpeg -i "$f" -c:a libmp3lame -b:a 128k "${f%.aiff}.mp3"; done. On Windows Command Prompt, use: for %f in (*.aiff) do ffmpeg -i "%f" -c:a libmp3lame -b:a 128k "%~nf.mp3". This is especially useful for large libraries of AIFF files that exceed the 1 GB browser limit.
Yes. FFmpeg handles all standard AIFF bit depths — including pcm_s16be (16-bit), pcm_s24be (24-bit), pcm_s32be (32-bit), and floating-point variants. The LAME encoder internally works with floating-point audio and will correctly down-convert the higher bit-depth PCM during encoding. The output MP3 will always be 16-bit equivalent, since the MP3 format does not support higher bit depths, but the LAME encoder uses the full dynamic range of the source during the psychoacoustic analysis.
Technical Notes
AIFF uses big-endian PCM encoding (hence codec names like pcm_s16be and pcm_s24be), a legacy of its development for Motorola 68000-based Macintosh hardware. When FFmpeg reads an AIFF file, it decodes this raw PCM stream into an internal floating-point representation before passing it to the LAME encoder. LAME (libmp3lame) is the de facto standard open-source MP3 encoder and produces compliant MPEG-1 Audio Layer III output at bitrates from 32 kbps to 320 kbps. The default 128 kbps constant bitrate (CBR) setting is a broadly compatible choice; for better quality-to-size ratio, consider using LAME's variable bitrate mode by replacing -b:a 128k with -q:a 2 (VBR quality scale 0–9, where 0 is best). One known limitation: AIFF supports audio at sample rates above 48 kHz (e.g., 88.2kHz or 96kHz hi-res audio), but MP3 is officially limited to 48 kHz maximum. FFmpeg and LAME will automatically resample higher sample rates down to 44.1 kHz or 48 kHz during encoding. If your AIFF source contains chapter markers or loop points (used in some game audio workflows), these will not be preserved in the MP3 output, as the MP3 container does not support such metadata structures.