Extract Audio from MPEG to AIFF — Free Online Tool
Extract audio from MPEG video files and save it as a lossless AIFF file, converting the compressed MP2 or MP3 audio stream into uncompressed PCM audio (pcm_s16be) suitable for professional audio workflows on macOS. This tool runs entirely in your browser — no upload required.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
MPEG files typically carry audio encoded with MP2 (MPEG Layer 2) or occasionally MP3 or AAC compression. Because AIFF is an uncompressed lossless format, FFmpeg fully decodes the compressed audio stream from the MPEG container and re-encodes it into 16-bit big-endian PCM audio (pcm_s16be), which is the standard AIFF codec. This is not a simple remux — the audio must be fully decoded and rewritten, though no video is processed at all (it is discarded). The result is an AIFF file whose audio quality is bounded by whatever fidelity survived the original lossy MPEG compression; uncompressed does not mean the original studio quality is restored, but all remaining audio information is preserved without any additional loss.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In the browser-based tool this runs via FFmpeg.wasm (WebAssembly) entirely locally; when running on your desktop, this calls your system-installed FFmpeg executable. |
-i input.mpeg
|
Specifies the input MPEG file. FFmpeg will detect whether it is an MPEG-1 or MPEG-2 program stream and identify all contained streams, including the compressed MP2 (or other) audio track to be extracted. |
-vn
|
Disables video output entirely. Since this is an audio extraction tool, the MPEG video stream (mpeg1video or mpeg2video) is discarded without being decoded, which significantly speeds up processing and ensures no video data is written into the AIFF output. |
-c:a pcm_s16be
|
Sets the audio codec to signed 16-bit big-endian PCM, which is the standard and expected codec inside an AIFF file. FFmpeg will fully decode the lossy MP2 audio from the MPEG source and re-encode it as uncompressed PCM samples in AIFF's native big-endian byte order. |
output.aiff
|
Defines the output filename and tells FFmpeg to write an AIFF container. The .aiff extension is recognized by macOS applications including Logic Pro, GarageBand, QuickTime Player, and Final Cut Pro as a standard uncompressed audio file. |
Common Use Cases
- Importing audio from archived MPEG broadcast recordings into Logic Pro or GarageBand on macOS, which natively prefer AIFF for editing sessions
- Extracting a music or voice track from a legacy MPEG-1 or MPEG-2 video (such as a VCD or DVD rip) to use as a clean audio asset in a post-production pipeline
- Converting MPEG audio to AIFF for use in professional audio software like Pro Tools or Final Cut Pro that expects uncompressed PCM audio for high-accuracy editing
- Archiving the audio portion of historical or broadcast MPEG recordings in an uncompressed format to prevent any further generational quality loss during future edits
- Preparing audio extracted from MPEG source material for waveform analysis or forensic audio review tools that require raw PCM input
- Stripping the audio from MPEG video files delivered by a broadcaster or legacy encoding system so it can be re-synced or remixed without re-encoding the video
Frequently Asked Questions
No — AIFF is lossless and uncompressed, but converting to it cannot recover quality that was already discarded during the original MPEG encoding. The MP2 audio inside an MPEG file is a lossy format, so any compression artifacts from that stage are permanently baked into the signal. What AIFF does guarantee is that no additional quality is lost during this conversion or in any subsequent re-export from your audio software, making it the right format for further editing.
MPEG files store audio as compressed MP2 or MP3 data, which can be 5 to 10 times smaller than uncompressed audio for a given duration. AIFF stores every audio sample as raw 16-bit PCM data with no compression whatsoever, so a one-hour MPEG file whose audio was encoded at 192 kbps MP2 will produce an AIFF of roughly 600 MB for the audio alone. This size increase is expected and reflects the shift from a compressed to an uncompressed container.
The output uses pcm_s16be — signed 16-bit big-endian PCM audio — which is the standard codec for AIFF files and what Apple's own tools (GarageBand, Logic Pro, QuickTime) expect when opening AIFF. The 'be' (big-endian) byte order is a defining characteristic of AIFF inherited from its development on Motorola-based Macintosh hardware and distinguishes it from WAV, which uses little-endian PCM.
Yes — FFmpeg preserves the original sample rate (commonly 44,100 Hz or 48,000 Hz in MPEG files) and the channel layout (mono or stereo) from the source audio stream. No resampling or downmixing occurs unless you add additional flags to the command. If your MPEG file has stereo MP2 audio at 48 kHz, the AIFF output will also be stereo at 48 kHz.
Replace pcm_s16be in the command with another supported AIFF codec to change the bit depth. For 24-bit audio use pcm_s24be, for 32-bit integer use pcm_s32be, and for 32-bit or 64-bit floating point use pcm_f32be or pcm_f64be respectively. For example: ffmpeg -i input.mpeg -vn -c:a pcm_s24be output.aiff. Note that since the source is lossy MP2 audio, increasing bit depth beyond 16-bit adds no real audio information — it can be useful for headroom in editing, but is not a quality improvement.
Yes, on Linux or macOS you can use a shell loop: for f in *.mpeg; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.mpeg}.aiff"; done. On Windows Command Prompt, use: for %f in (*.mpeg) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aiff". The browser-based tool processes one file at a time, so use the displayed FFmpeg command locally for batch operations, especially for collections of large MPEG files exceeding 1 GB.
Technical Notes
MPEG containers (both MPEG-1 and MPEG-2 program streams) typically multiplex MP2 audio, occasionally MP3 or AAC. The -vn flag ensures FFmpeg ignores the MPEG video stream entirely, so no video decoding occurs — this keeps conversion fast even for long files. AIFF does not support metadata tags in the same rich way that MP4 or Ogg containers do; basic metadata like track name may not carry over from the MPEG file, and most MPEG program streams carry minimal metadata anyway. AIFF also does not support multiple audio tracks, chapters, or subtitles, none of which are typically present in standard MPEG files. The pcm_s16be codec produces CD-standard 16-bit audio, which is appropriate for source material originally compressed at typical MPEG broadcast bitrates (128–256 kbps MP2). The output AIFF will be compatible with macOS QuickTime Player, Logic Pro, GarageBand, Final Cut Pro, Audacity, and most professional digital audio workstations. Files with unusual MPEG audio streams (e.g., LPCM audio in a DVD MPEG-2 transport stream) will still be decoded correctly and written to AIFF without any additional flags.