Convert ALAC to WAV — Free Online Tool
Convert ALAC audio files (.m4a) to WAV format by decoding Apple's lossless compressed audio into uncompressed PCM (16-bit signed little-endian) — producing a universally compatible audio file with no quality loss. Ideal for moving Apple-ecosystem audio into professional workflows, DAWs, and broadcast environments that expect raw PCM data.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your ALAC file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
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 (Apple Lossless Audio Codec) stores audio as losslessly compressed PCM data inside an MPEG-4 container (.m4a). During this conversion, FFmpeg decodes the ALAC-compressed bitstream back to its original uncompressed PCM samples, then writes them into a WAV container using the pcm_s16le codec — 16-bit signed integers in little-endian byte order. Because ALAC is lossless, the decoded audio is mathematically identical to the original uncompressed signal, meaning the resulting WAV is a perfect reconstruction with zero quality degradation. The trade-off is file size: WAV stores every sample without compression, so the output will be noticeably larger than the ALAC source. Metadata tags (artist, album, title) stored in the MPEG-4 container are not preserved in the WAV output, as WAV has only limited and inconsistently supported metadata capabilities.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no files leave your device. |
-i input.m4a
|
Specifies the input file — an MPEG-4 container (.m4a) carrying an ALAC-encoded audio stream. FFmpeg reads and demuxes the container before passing the ALAC bitstream to the decoder. |
-c:a pcm_s16le
|
Decodes the ALAC audio and re-encodes it as signed 16-bit little-endian PCM — the standard uncompressed audio codec used in WAV files. This is the format expected by the vast majority of audio software, hardware, and broadcast systems. |
output.wav
|
Defines the output file. The .wav extension tells FFmpeg to wrap the raw PCM audio in a RIFF/WAV container, which is the standard packaging for PCM audio on Windows, macOS, and Linux platforms. |
Common Use Cases
- Importing Apple Music rips or iTunes-purchased ALAC tracks into a DAW like Pro Tools, Logic Pro, or Ableton Live, which may require flat PCM WAV files for reliable editing and plugin compatibility.
- Preparing audio for broadcast or mastering delivery, where studios and post-production houses require uncompressed 16-bit PCM WAV files as the standard interchange format.
- Using ALAC tracks as source audio in video editing software (Adobe Premiere, DaVinci Resolve) that handles WAV more reliably than MPEG-4 audio-only containers.
- Converting an ALAC music library to WAV for playback on hardware devices — such as DJ controllers, hardware samplers, or digital mixers — that support WAV but not ALAC.
- Archiving or sharing audio with collaborators on non-Apple platforms where ALAC decoding support cannot be assumed, but WAV playback is universally available.
- Running audio analysis or processing tools (spectrum analyzers, loudness meters, batch processors) that require raw PCM input and do not support ALAC-in-M4A containers.
Frequently Asked Questions
No — this conversion is fully lossless in terms of audio fidelity. ALAC is a lossless codec, meaning it stores a perfect compressed representation of the original PCM audio. When FFmpeg decodes ALAC and writes it to WAV as pcm_s16le, it reconstructs the exact same 16-bit sample values that were in the source. The resulting WAV file is bit-for-bit equivalent to the uncompressed audio the ALAC file encodes.
ALAC achieves roughly 40–60% compression compared to raw PCM without any quality loss, similar to how ZIP compresses data. WAV with pcm_s16le stores every audio sample as a raw integer with no compression at all — so a 30-minute ALAC file that might be 150MB could expand to 250–300MB as a WAV. This is expected and normal; you are not adding data, just removing the compression layer.
Generally no. ALAC files stored in MPEG-4 containers (.m4a) can carry rich metadata including artist, album, genre, artwork, and track numbers. WAV has a limited metadata standard (the INFO chunk and the newer ID3 chunk) that is inconsistently supported across software. FFmpeg may write some basic tags, but album art will not transfer, and many WAV-reading applications will simply ignore any embedded tags. If metadata preservation is critical, consider keeping the ALAC original as your archive copy.
Replace pcm_s16le with pcm_s24le in the command: ffmpeg -i input.m4a -c:a pcm_s24le output.wav. This is useful if your ALAC source was recorded or mastered at 24-bit depth, since pcm_s16le would truncate those extra bits. If the ALAC source is only 16-bit, using pcm_s24le will upsample the bit depth without adding real information. Check your source file's bit depth first using ffprobe.
Yes. On macOS or Linux, you can run: for f in *.m4a; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.m4a}.wav"; done in your terminal. On Windows (Command Prompt), use: for %f in (*.m4a) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This processes each ALAC file in the current directory and outputs a matching WAV file with the same base filename.
Yes, chapters will be lost. ALAC files in MPEG-4 containers can store chapter markers, which is useful for audiobooks or long-form recordings. WAV does not have a native chapter or cue-point structure that FFmpeg writes by default. If chapter markers are important, you would need to either keep the ALAC file or export to a format like FLAC or M4A that supports chapter metadata.
Technical Notes
The FFmpeg codec pcm_s16le — 'PCM, signed 16-bit, little-endian' — is the WAV default and the most universally compatible PCM variant. It matches the CD audio standard (16-bit, 44100 Hz stereo), making it the safest choice for general-purpose compatibility. However, if the ALAC source was encoded at 24-bit depth (common for studio recordings and high-resolution downloads), converting to pcm_s16le introduces a bit-depth reduction from 24 to 16 bits, which is a form of precision loss — not lossy compression, but a quantization reduction. In that case, using pcm_s24le preserves the full depth. The sample rate is passed through unchanged by default; a 96kHz ALAC file will produce a 96kHz WAV. WAV files using pcm_s16le have a theoretical maximum file size of 4GB due to the 32-bit chunk size field in the RIFF header — for very long or high-sample-rate recordings, this limit may be relevant. ALAC's MPEG-4 container does not support multiple audio tracks, so there is no multi-track handling to worry about. No video stream is present in an ALAC .m4a file, so no video-related flags are needed in the command.