Convert MP3 to AIFC — Free Online Tool
Convert MP3 audio files to AIFC format using the PCM signed 16-bit big-endian codec, transforming lossy compressed audio into a professional-grade container suitable for Apple and broadcast workflows. This tool decodes your MP3 in the browser using FFmpeg.wasm and re-encodes it as uncompressed PCM audio wrapped in the AIFC container — no upload required.
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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
MP3 files store audio using the MPEG Audio Layer III lossy compression algorithm, which permanently discards audio data to achieve small file sizes. During conversion to AIFC, FFmpeg fully decodes the MP3 bitstream back to raw PCM audio samples, then re-encodes those samples as 16-bit signed big-endian PCM (pcm_s16be) inside an AIFC container. Because the MP3 was lossy to begin with, the resulting AIFC file contains uncompressed PCM audio — but the audio quality ceiling is capped at whatever survived the original MP3 encoding. No further audio data is lost in this process; AIFC simply stores the decoded samples without any additional compression. The AIFC container uses big-endian byte ordering, which is native to the format's Apple and professional audio heritage, and the resulting file will be significantly larger than the source MP3.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg application, which is running here as a WebAssembly binary (FFmpeg.wasm) compiled to run entirely inside your browser without any server-side processing. |
-i input.mp3
|
Specifies the input file as an MP3, telling FFmpeg to read and decode the MPEG Audio Layer III bitstream from this file as the audio source for the conversion. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, which is the standard uncompressed audio encoding for the AIFC container — converting the lossy decoded MP3 audio into uncompressed PCM samples stored in Apple's native big-endian byte order. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps, though this parameter has no practical effect on uncompressed PCM codecs like pcm_s16be since the actual bitrate of PCM audio is fixed by the sample rate and bit depth rather than a compression target. |
output.aifc
|
Defines the output filename and tells FFmpeg to write the result into an AIFC container, which FFmpeg infers from the .aifc file extension and which is the Apple-originated format designed to hold PCM and compressed audio in a chunk-based structure. |
Common Use Cases
- Importing audio into Apple Logic Pro or Final Cut Pro, which prefer AIFF/AIFC over MP3 for timeline editing to avoid re-encoding artifacts during mixing
- Delivering audio assets to broadcast or post-production facilities that require uncompressed PCM in an Apple-compatible container rather than compressed MP3 files
- Preparing MP3 music tracks for use in older Apple hardware or software samplers that only accept AIFF or AIFC as source formats
- Archiving MP3 audio into an uncompressed PCM container so downstream audio editing preserves the decoded signal without stacking additional lossy encoding passes
- Converting a finished MP3 podcast episode into AIFC for ingestion into a DAW for mastering, loudness normalization, or adding broadcast-compliant metadata
- Providing audio files to clients or studios on macOS who require AIFC as their standard interchange format for sound design pipelines
Frequently Asked Questions
No — converting from MP3 to AIFC does not recover any audio quality that was lost during the original MP3 encoding. The MP3 codec permanently discards frequency information during compression, and those losses cannot be reversed by re-encoding into an uncompressed format. What you get is the decoded MP3 audio stored as uncompressed PCM inside AIFC, which means no further quality degradation will occur during subsequent editing or processing in a DAW. Think of it as 'freezing' the quality at whatever level the MP3 had.
MP3 uses aggressive lossy compression — a 128 kbps MP3 stores audio at a fraction of what uncompressed PCM requires. AIFC with pcm_s16be stores 16 bits per sample for every sample at the full sample rate (typically 44,100 samples per second per channel), which works out to approximately 10 MB per minute for stereo audio. A 3-minute MP3 at 128 kbps might be around 2.9 MB, while the equivalent AIFC file will be roughly 30 MB. This size increase is expected and reflects the removal of compression, not any duplication of data.
pcm_s16be stands for PCM signed 16-bit big-endian — it stores each audio sample as a 16-bit integer in big-endian byte order, which means the most significant byte comes first. This is the native uncompressed audio format associated with the AIFF/AIFC family of containers, originating from Apple's use of Motorola processors which used big-endian architecture. It is the most universally compatible codec choice within the AIFC container and is supported by virtually all professional audio software on macOS. AIFC technically supports compressed codecs like A-law and mu-law, but pcm_s16be is the standard choice for high-quality lossless storage.
MP3 files store metadata using ID3 tags, while AIFC uses its own chunk-based metadata system derived from the AIFF specification. FFmpeg will attempt to map common ID3 fields such as title, artist, and album into AIFC-compatible metadata chunks during the conversion, but not all ID3 fields have direct AIFC equivalents and some metadata may be dropped or not recognized by all software. If precise metadata preservation is critical, you should verify the output file's tags in your audio application after conversion and re-enter any missing fields manually.
You can replace pcm_s16be in the command with other PCM variants that AIFC supports to change the bit depth or encoding. For example, use -c:a pcm_s24be for 24-bit audio (giving more headroom for editing), -c:a pcm_s32be for 32-bit integer, or -c:a pcm_f32be for 32-bit floating point — all still in big-endian format. The command would look like: ffmpeg -i input.mp3 -c:a pcm_s24be output.aifc. Note that increasing the bit depth beyond 16-bit will not recover quality lost in the original MP3, but 32-bit float is often preferred inside DAWs for internal processing to avoid clipping.
Yes — on the command line you can use a shell loop to process multiple files at once. On Linux or macOS, the command would be: for f in *.mp3; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mp3}.aifc"; done. On Windows Command Prompt you can use: for %f in (*.mp3) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc". This is particularly useful when you have more than a few files to convert, or when working with files larger than 1 GB that cannot be processed in the browser-based tool.
Technical Notes
The AIFC format (Audio Interchange File Format Compressed) is a superset of Apple's AIFF format and supports both uncompressed PCM and several compressed audio encodings including A-law (pcm_alaw) and mu-law (pcm_mulaw). When converting from MP3, FFmpeg uses libmp3lame to decode the input and writes the resulting PCM samples into AIFC's chunk-based container structure using big-endian byte ordering. The -b:a flag in this conversion context applies to the output but has no meaningful effect on uncompressed PCM codecs like pcm_s16be, since PCM bitrate is determined by sample rate, channel count, and bit depth rather than a compression target. The output sample rate will match the source MP3's sample rate (commonly 44,100 Hz or 48,000 Hz) unless you explicitly specify -ar in the command. AIFC files use the .aifc or .aif extension and are natively readable by macOS QuickTime, Logic Pro, GarageBand, and most professional DAWs. One known limitation is that AIFC does not support multiple audio tracks, chapters, or embedded subtitles, so this is strictly a single stereo or mono audio container. If your MP3 has joint stereo encoding, FFmpeg will decode it to standard stereo PCM in the output.