Convert MPEG to AIFF — Free Online Tool
Convert MPEG video files to AIFF audio by extracting and decoding the MP2 or AAC audio track into uncompressed PCM audio stored in Apple's lossless AIFF container. Ideal for recovering broadcast-quality audio from legacy MPEG recordings for use in professional audio workflows on macOS.
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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
MPEG files typically carry audio encoded with MP2 (MPEG Layer 2) or occasionally AAC compression alongside MPEG-1 or MPEG-2 video. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio track, converting it to 16-bit big-endian PCM (pcm_s16be) — the raw, uncompressed audio format native to AIFF. The result is a lossless AIFF file containing the full audio content of the original MPEG, free from any further lossy compression. Because the MPEG audio was already lossy, the AIFF output captures the decoded audio at its current fidelity — it does not recover detail lost during the original MPEG encoding, but it does prevent any additional generation loss.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion both in the browser (via FFmpeg.wasm) and on the desktop command line. |
-i input.mpeg
|
Specifies the input MPEG file. FFmpeg automatically detects whether the container holds MPEG-1 or MPEG-2 video and identifies the audio codec (typically MP2, but possibly AAC or AC-3) from the stream headers. |
-c:a pcm_s16be
|
Decodes the compressed MPEG audio (MP2 or AAC) and re-encodes it as 16-bit signed big-endian PCM — the uncompressed audio format native to AIFF. Big-endian byte order is required by the AIFF specification, distinguishing it from WAV's little-endian PCM. |
output.aiff
|
Defines the output filename and instructs FFmpeg to write an AIFF container. The .aiff extension causes FFmpeg to apply the correct AIFF container format, which wraps the pcm_s16be audio stream in Apple's standard uncompressed audio interchange structure. |
Common Use Cases
- Extracting the audio commentary or narration from a legacy MPEG broadcast recording for re-editing in Logic Pro or Final Cut Pro on macOS
- Archiving the audio from old MPEG-1 or MPEG-2 home recordings into a lossless format before the source media degrades further
- Preparing audio from DVD-compatible MPEG files for import into professional DAWs like Pro Tools, which prefer uncompressed AIFF over MP2 or MP3
- Pulling a music performance or live event recording captured in MPEG format into an uncompressed AIFF for sample-accurate editing without transcoding artifacts
- Converting MPEG news broadcast audio to AIFF for forensic or legal audio analysis, where uncompressed PCM is required by the review workflow
- Separating the audio track from an MPEG file received from a broadcast or streaming archive so it can be transcribed or captioned using macOS-native tools
Frequently Asked Questions
No — converting from MPEG to AIFF does not restore quality lost during the original MPEG encoding. MPEG audio (typically MP2 or AAC) is lossy, meaning some audio information was discarded when the MPEG was first created. The AIFF file stores the decoded result of that lossy audio as uncompressed PCM, which accurately preserves what remains but cannot recover what was lost. The benefit of AIFF here is preventing any further quality degradation in downstream processing.
The command defaults to pcm_s16be (16-bit signed big-endian PCM) because it matches CD-quality audio and is the most universally compatible PCM format for AIFF playback on macOS and in most DAWs. Since MPEG audio sources — particularly MP2 at broadcast bitrates — rarely contain meaningful detail beyond 16-bit resolution, the higher bit depths (24-bit or 32-bit) would increase file size without a perceptible quality benefit. If you need 24-bit output for professional mixing headroom, you can change the codec flag to -c:a pcm_s24be in the command.
Significantly larger — AIFF stores uncompressed PCM audio, while MPEG audio (MP2 at 192 kbps, for example) is heavily compressed. A one-hour MPEG with 192 kbps MP2 audio will produce an AIFF of roughly 600 MB at 16-bit/44.1 kHz stereo, compared to the approximately 86 MB that the same audio occupies as MP2. This size increase is expected and is the cost of storing audio in a lossless, uncompressed format suitable for professional editing.
MPEG files carry minimal standardized metadata, and AIFF has limited metadata support compared to formats like FLAC or WAV with ID3 tags. FFmpeg will attempt to copy any available metadata markers during conversion, but in practice most MPEG recordings contain little or no embedded metadata beyond technical stream properties. Do not rely on this conversion to preserve titles, descriptions, or timestamps from broadcast MPEG sources.
Replace pcm_s16be in the command with your desired codec: use pcm_s24be for 24-bit output or pcm_s32be for 32-bit integer PCM. For example: ffmpeg -i input.mpeg -c:a pcm_s24be output.aiff. You can also use pcm_f32be or pcm_f64be for 32-bit or 64-bit floating-point PCM, which some DAWs use internally for mixing. Unless your downstream workflow specifically requires a higher bit depth, pcm_s16be is sufficient for audio originally encoded as MP2 or AAC.
Yes. On macOS or Linux, you can use a shell loop to process multiple files: for f in *.mpeg; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mpeg}.aiff"; done. On Windows Command Prompt, use: for %f in (*.mpeg) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This is especially useful when processing large collections of archived MPEG broadcast recordings, and is one reason the FFmpeg command is displayed on this page — the browser tool handles files up to 1 GB, but for bulk or larger conversions, running FFmpeg locally is more practical.
Technical Notes
MPEG containers (.mpeg, .mpg) primarily support MPEG-1 and MPEG-2 video alongside MP2 audio, though some MPEG-2 program streams also carry AAC or AC-3 audio. FFmpeg reliably detects the audio codec in use and decodes it to PCM regardless of whether the source is MP2 or AAC. The output pcm_s16be codec uses big-endian byte ordering, which is native to the AIFF format and required by the Apple specification — this differs from WAV, which uses little-endian PCM (pcm_s16le). AIFF does not support chapters, subtitle streams, or multiple audio tracks, but since MPEG files also lack multiple audio track support in most real-world encodings, this is rarely a limitation. One notable edge case: some MPEG-2 program streams (common in DVD-ripped content) may contain AC-3 (Dolby Digital) audio rather than MP2; FFmpeg handles this transparently, but the resulting AIFF will be stereo or mono depending on how FFmpeg downmixes the surround channels by default. If channel layout preservation matters, add explicit channel mapping flags to the command. There is no meaningful sample rate conversion during this process unless the source MPEG contains a non-standard sample rate — most broadcast MPEG files use 44.1 kHz or 48 kHz, both of which AIFF supports natively.