Convert MPEG to AIF — Free Online Tool
Extract and convert audio from MPEG video files into AIF format, producing uncompressed PCM audio stored in Apple's Audio Interchange File Format. This tool strips the MP2 or AAC audio stream from your MPEG file and re-encodes it as 16-bit big-endian PCM, giving you a lossless, uncompressed AIF file ideal for use in professional Mac-based audio workflows.
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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
MPEG files typically carry audio encoded in MP2 (MPEG-1 Audio Layer II), MP3, or AAC — all lossy compressed formats that discard audio data to reduce file size. During this conversion, FFmpeg demuxes the MPEG container, extracts the compressed audio stream, decodes it fully to raw PCM in memory, and then re-encodes it as 16-bit big-endian PCM wrapped in an AIF container. The video stream is discarded entirely since AIF is a pure audio format. Because the source audio was already lossy (MP2 at 192k is the MPEG default), the resulting AIF file is technically lossless from this point forward — no further generation loss will occur in subsequent editing — but it cannot recover the detail that the original MPEG compression discarded. The output file will be substantially larger than the MPEG source.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg media processing tool. In the browser version, this runs as a WebAssembly binary (FFmpeg.wasm) entirely within your browser — no server is involved. |
-i input.mpeg
|
Specifies the input MPEG file. FFmpeg will demux the MPEG container, making both the MPEG-1/2 video stream and the compressed audio stream (typically MP2) available for processing. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM — the standard uncompressed audio encoding used in AIF files. FFmpeg decodes the lossy MP2 audio from the MPEG and re-encodes it as raw PCM samples in big-endian byte order, which is what the AIF format expects. |
output.aif
|
Defines the output filename and format. The .aif extension tells FFmpeg to wrap the PCM audio data in an Audio Interchange File Format container, the Apple-native uncompressed audio format compatible with Logic Pro, GarageBand, Final Cut Pro, and most professional Mac audio tools. |
Common Use Cases
- Importing audio from archival MPEG broadcast recordings into Logic Pro or Final Cut Pro for editing on a Mac, where AIF is the native preferred format
- Extracting the audio track from a DVD-compatible MPEG file to hand off to a sound designer who works in an Apple-centric studio pipeline
- Recovering a clean, uncompressed audio stem from legacy MPEG content before re-encoding to a modern format like FLAC or AAC, to avoid accumulating multiple generations of lossy compression
- Preparing audio from MPEG news broadcast clips for ingestion into a professional audio workstation that requires uncompressed AIF input
- Archiving the audio portion of old MPEG home video files in a non-compressed format to ensure long-term playback compatibility without codec dependency
- Stripping and converting the MP2 audio track from a streaming MPEG capture into AIF for use as a source asset in a podcast editing project on macOS
Frequently Asked Questions
No — the AIF output will sound identical to the audio in the source MPEG, not better. MPEG files use lossy compression (typically MP2 or AAC), which permanently removes audio information during the original encoding. Converting to AIF decodes that lossy stream and stores it uncompressed, which prevents any further quality loss in future edits or re-encodes, but it cannot restore the detail that was discarded when the MPEG was first created.
AIF stores audio as uncompressed PCM — every sample is written as raw data with no compression applied. A standard 16-bit stereo AIF file at 44.1kHz uses roughly 10MB per minute of audio. MPEG's MP2 audio at 192k uses approximately 1.4MB per minute, so an AIF file will typically be around 7–8 times larger than the equivalent MPEG audio duration. This size increase is expected and is the nature of lossless uncompressed storage.
The video track is completely discarded. AIF is a pure audio container and cannot store video streams, so FFmpeg extracts only the audio from the MPEG file and ignores the MPEG-1 or MPEG-2 video entirely. If you need to keep the video, you would need a different output format such as MOV or MP4.
Yes. FFmpeg's MP2 decoder is mature and accurate, and MP2 is a well-specified codec with no proprietary quirks. The decoded PCM data written into the AIF file will be a faithful representation of what the MP2 stream contains. The only quality ceiling is the original MP2 bitrate — typically 192k in standard MPEG files — which is the rate at which audio detail was discarded at source.
Yes. The default command uses '-c:a pcm_s16be' for 16-bit big-endian PCM, which is the AIF standard. You can substitute '-c:a pcm_s24be' for 24-bit or '-c:a pcm_s32be' for 32-bit PCM if your downstream application requires higher bit depth. However, since the source is lossy MP2 audio (typically encoded from 16-bit source material), increasing the bit depth will not recover any additional audio detail — it will simply store the same decoded audio in a wider container.
Yes. On Linux or macOS you can loop over files in a directory using a shell one-liner: 'for f in *.mpeg; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mpeg}.aif"; done'. On Windows Command Prompt you can use 'for %f in (*.mpeg) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif"'. This is especially useful for processing large collections of MPEG broadcast recordings that exceed the 1GB browser tool limit.
Technical Notes
AIF uses big-endian byte ordering for its PCM data, which is why the codec specified is 'pcm_s16be' (signed 16-bit big-endian) rather than the little-endian 'pcm_s16le' used in WAV files. This matters if you are processing the raw audio data programmatically. MPEG containers do not carry extensive metadata — typically no ID3 tags or chapter markers — so there is little metadata to preserve or lose in this conversion; the AIF output will generally have no embedded title, artist, or album fields. AIF itself has limited metadata support compared to formats like FLAC or MP4. If your MPEG source uses the less common AAC audio codec variant rather than MP2, FFmpeg will decode it correctly and the output AIF will be identical in structure. The sample rate of the output AIF will match whatever sample rate was used in the MPEG source (commonly 44.1kHz or 48kHz); no sample rate conversion is applied by default. AIF has no inherent file size limit that would affect normal use, but very long MPEG files converted to uncompressed AIF may produce files that exceed the storage expectations of some users unfamiliar with uncompressed audio sizes.