Convert ALAC to MP3 — Free Online Tool
Convert ALAC (.m4a) files to MP3 using the LAME encoder directly in your browser — no upload required. This conversion trades Apple's lossless compression for widely compatible lossy MP3, dramatically reducing file size while retaining perceptually acceptable audio quality for most listening scenarios.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your ALAC 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
ALAC stores audio as lossless PCM data inside an MPEG-4 container, meaning every sample from the original recording is preserved. During this conversion, FFmpeg decodes the ALAC stream back to raw PCM, then re-encodes it using the libmp3lame encoder at 128 kbps CBR, producing an MP3 file. Because ALAC is lossless and MP3 is lossy, this is a one-way lossy transcode — the psychoacoustic compression applied by LAME permanently discards audio data deemed imperceptible to the human ear. The result is a file roughly 5–10x smaller than the original ALAC, encoded in a format natively supported by virtually every device, browser, and media player.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser version, this runs via FFmpeg.wasm, a WebAssembly port that executes the same conversion logic client-side without sending your ALAC file to any server. |
-i input.m4a
|
Specifies the input file — your ALAC audio stored in an MPEG-4 container with the .m4a extension. FFmpeg reads the container, identifies the ALAC audio stream, and decodes it to raw PCM before passing it to the encoder. |
-c:a libmp3lame
|
Selects the LAME MP3 encoder for the audio stream. LAME is the industry-standard open-source MP3 encoder, used here to re-encode the decoded ALAC PCM data into the lossy MP3 format that every major device and platform supports natively. |
-b:a 128k
|
Sets the MP3 audio bitrate to 128 kbps Constant Bit Rate. This is the conventional default that balances file size and perceptual quality — suitable for casual listening. Increase this value (e.g., 320k) when transcoding from a lossless ALAC source to preserve more of the original detail. |
output.mp3
|
Defines the output filename and format. The .mp3 extension tells FFmpeg to write an MPEG Audio Layer III file using the ID3 tag structure for metadata, replacing the MPEG-4 container that held the original ALAC data. |
Common Use Cases
- Sharing a high-quality ALAC music rip with a friend whose Android device or non-Apple media player doesn't support the ALAC codec
- Reducing the file size of a large ALAC audiophile library to fit onto a storage-limited device or USB drive
- Uploading Apple Music exports or iTunes purchases (in ALAC form) to platforms like SoundCloud or Mixcloud that require MP3 input
- Preparing podcast interview recordings that were captured in ALAC for distribution, since all podcast hosting platforms expect MP3
- Converting ALAC audiobook files purchased from Apple into MP3 so they can be played in non-Apple audiobook apps or car stereos
- Creating a portable, smaller MP3 copy of an ALAC master archive to use as a listening copy on everyday devices while keeping the lossless original
Frequently Asked Questions
Yes — this is an irreversible lossy conversion. ALAC is a lossless format, meaning it contains every bit of the original audio signal. MP3 uses psychoacoustic modeling to discard audio information considered inaudible, so some detail is permanently lost. At 128 kbps (the default used here) most listeners won't notice a difference on typical headphones or speakers, but audiophiles comparing against the original ALAC on high-end equipment will likely detect subtle degradation, particularly in high frequencies and complex passages.
ALAC achieves compression without discarding any audio data, typically producing files around 3–5 MB per minute of audio. MP3 at 128 kbps achieves roughly 1 MB per minute by permanently removing audio data through lossy compression. For a typical 4-minute song, you can expect the ALAC source to shrink from around 15–20 MB down to approximately 4 MB as an MP3 — a size reduction of 75–80%.
Partially. ALAC files store metadata as iTunes-style MP4 tags, while MP3 uses the ID3 tag standard. FFmpeg will attempt to map common fields such as title, artist, album, and track number from the ALAC container to ID3 tags in the output MP3, and this usually succeeds for standard fields. However, more obscure or Apple-specific metadata fields may not transfer, and embedded artwork conversion can be inconsistent — you may want to verify tags in a tool like MusicBrainz Picard after conversion.
Change the `-b:a 128k` value to a higher bitrate. For example, `-b:a 320k` produces the highest standard MP3 quality and is recommended if you're archiving music or need the best possible fidelity from the lossless source. A value of `-b:a 192k` is a good middle ground — noticeably higher quality than 128k with a moderate file size increase. Since you're transcoding from a lossless ALAC source, there's no penalty for encoding at high bitrates; the LAME encoder has full access to the original audio.
Yes, the displayed command can be adapted for batch processing in a terminal. On Linux or macOS, you can run: `for f in *.m4a; do ffmpeg -i "$f" -c:a libmp3lame -b:a 128k "${f%.m4a}.mp3"; done`. On Windows (PowerShell), use: `Get-ChildItem *.m4a | ForEach-Object { ffmpeg -i $_.FullName -c:a libmp3lame -b:a 128k ($_.BaseName + '.mp3') }`. The in-browser tool handles one file at a time, so the FFmpeg command is the recommended route for large libraries.
Yes — that's one of the primary reasons to make this conversion. ALAC playback requires explicit codec support, which is absent on many Android devices, non-Apple smart TVs, car stereos, and older media players. MP3 is supported by virtually every audio playback device manufactured in the last two decades, making it the safest format choice for broad compatibility.
Technical Notes
ALAC files use the .m4a file extension and are stored in an MPEG-4 container, which can cause confusion since AAC audio files share the same extension. FFmpeg correctly identifies the internal ALAC codec regardless of extension and handles the decode cleanly. The libmp3lame encoder used in this command is the open-source LAME library, widely regarded as producing the highest-quality MP3 output available. By default this tool encodes at 128 kbps CBR (Constant Bit Rate); for audiophile-grade output from a lossless source, consider using VBR mode instead by replacing `-b:a 128k` with `-q:a 2` in the command, which targets roughly 190 kbps variable bitrate and is often preferred for quality-sensitive archiving. One important limitation: MP3 does not support chapter markers, so any chapter data present in the ALAC file will be dropped. Additionally, MP3's maximum supported sample rate is 48 kHz — ALAC files recorded at higher sample rates (such as 96 kHz hi-res audio) will be downsampled automatically by FFmpeg during the transcode.