Convert MP3 to AIFF — Free Online Tool
Convert MP3 files to AIFF format directly in your browser, decoding the lossy MP3 audio and repackaging it as uncompressed PCM (pcm_s16be) for use in Apple-native workflows, professional audio editors, and macOS applications. AIFF's raw 16-bit big-endian PCM encoding makes it immediately compatible with Logic Pro, GarageBand, and Final Cut Pro without any additional transcoding step.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
MP3 uses lossy compression via the MPEG Audio Layer III codec (libmp3lame), discarding audio data deemed perceptually inaudible to achieve small file sizes. During this conversion, FFmpeg fully decodes the MP3 bitstream back to raw PCM audio samples, then re-encodes those samples as 16-bit big-endian PCM (pcm_s16be) wrapped in an AIFF container. Importantly, this is a lossy-to-lossless conversion, meaning the AIFF output will be uncompressed and perfectly preserved from this point forward — but any quality lost during the original MP3 encoding cannot be recovered. The output file will be significantly larger than the source MP3 because AIFF stores every audio sample without compression, typically producing files 8–10 times larger at 128kbps source quality.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles the MP3 decoding and AIFF encoding pipeline in this conversion. In the browser-based tool, this runs as FFmpeg.wasm compiled to WebAssembly. |
-i input.mp3
|
Specifies the input file — your source MP3 audio. FFmpeg reads and fully decodes the MPEG Audio Layer III bitstream from this file into raw PCM samples for processing. |
-c:a pcm_s16be
|
Sets the audio codec for the output to 16-bit big-endian signed PCM, which is the standard uncompressed audio encoding used inside AIFF files. This produces CD-quality, fully uncompressed audio data compatible with Logic Pro, GarageBand, and other Apple audio tools. |
output.aiff
|
Defines the output filename with the .aiff extension, which tells FFmpeg to wrap the pcm_s16be audio stream in an Audio Interchange File Format container — Apple's native uncompressed audio format optimized for macOS-based professional audio applications. |
Common Use Cases
- Importing an MP3 music track into Logic Pro or GarageBand, which work natively and most efficiently with AIFF files for timeline editing and mixing
- Preparing an MP3 sound effect or voiceover for use in a Final Cut Pro project, where AIFF avoids additional real-time decoding overhead during playback
- Archiving a collection of MP3 recordings in AIFF format to prevent further quality degradation from any future re-encoding steps in an audio production pipeline
- Sending audio files to a mastering engineer or studio that requires uncompressed AIFF deliverables as a standard intake format on macOS-based systems
- Loading an MP3 sample into hardware or software samplers (such as the Native Instruments MPC or Kontakt) that prefer or require uncompressed AIFF or WAV sample files
- Converting a podcast episode or interview recorded in MP3 to AIFF so it can be imported into an audio restoration tool that operates on uncompressed source material
Frequently Asked Questions
No — the AIFF output will be uncompressed and lossless from this point forward, but it cannot restore quality that was discarded when the MP3 was originally encoded. Think of it like enlarging a low-resolution JPEG: the resulting file is larger and won't degrade further, but the original compression artifacts are baked in. The primary benefit is preventing any additional quality loss in future processing or re-encoding steps.
MP3 at 128kbps uses aggressive lossy compression to shrink audio data. AIFF with pcm_s16be stores every audio sample as a raw 16-bit integer — no compression whatsoever. A 4-minute MP3 at 128kbps might be around 3.8MB, while the equivalent AIFF will be approximately 40MB. This size increase is expected and is the nature of converting from compressed to uncompressed audio storage.
pcm_s16be stands for PCM (Pulse-Code Modulation) audio stored as signed 16-bit integers in big-endian byte order — the native format for AIFF files and standard CD-quality audio. Since MP3 files are typically derived from 16-bit or 24-bit sources but encoded at bitrates that don't justify higher bit depths, 16-bit output is almost always sufficient. If your downstream workflow specifically requires 24-bit or 32-bit AIFF, you can modify the FFmpeg command to use pcm_s24be or pcm_s32be, but this won't add meaningful audio information beyond what the MP3 contained.
MP3 files commonly carry ID3 metadata tags, while AIFF uses its own chunk-based metadata structure. FFmpeg will attempt to map common ID3 fields (title, artist, album, year, track number) to AIFF-compatible metadata during conversion, though some extended or custom ID3 tags may not transfer cleanly. Applications like iTunes and Logic Pro can read and display standard AIFF metadata fields correctly after conversion.
To use a higher bit depth in the AIFF output, replace pcm_s16be in the command with pcm_s24be (24-bit) or pcm_s32be (32-bit): ffmpeg -i input.mp3 -c:a pcm_s24be output.aiff. You can also use floating-point codecs like pcm_f32be if required by a specific application. Keep in mind that the sample rate will be inherited from the source MP3 by default; to resample to 44100 Hz explicitly, add -ar 44100 before the output filename.
Yes. On macOS or Linux, you can use a shell loop: for f in *.mp3; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mp3}.aiff"; done. On Windows Command Prompt, use: for %f in (*.mp3) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This processes each MP3 in the current directory and saves a corresponding AIFF file with the same base filename. The browser-based tool processes one file at a time, so the FFmpeg command is especially useful for bulk conversions of large collections.
Technical Notes
The conversion from MP3 to AIFF involves full decode-then-encode processing — there is no stream copying (remuxing) possible here because the containers and codecs are fundamentally different. FFmpeg decodes the MP3 using its built-in MPEG audio decoder, producing an intermediate PCM representation in memory, then writes that PCM data into an AIFF container using the pcm_s16be codec, which stores samples as big-endian 16-bit signed integers — the byte order that Apple's AIFF specification requires. The output sample rate and channel count mirror the source MP3 exactly (commonly 44100 Hz stereo). AIFF does not support multiple audio tracks, embedded subtitles, or chapters, but neither does MP3, so no data is lost on those fronts. One notable limitation: MP3 files encoded with a joint stereo or mid-side stereo mode will decode to standard stereo PCM correctly, but any gapless playback metadata (such as LAME encoder delay tags) is discarded in the AIFF output, which could introduce a brief silence artifact at the start or end of tracks in gapless album playback scenarios. For professional workflows requiring the highest fidelity from an MP3 source, using pcm_s24be or pcm_f32be as the output codec provides headroom for any downstream processing even though the perceptual ceiling is set by the original MP3 encode.