MPEG-2 to AIFF Converter

Convert MPEG-2 files to AIFF format online. Free, fast, no watermarks.

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Supports: MPEG2

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Extract AIFF Audio from an MPEG-2 File

A .mpeg2 file is an MPEG program stream — the packaging DVD-Video, tuner captures and broadcast archives use — and its soundtrack is almost always MP2 or Dolby Digital (AC-3). This tool decodes that soundtrack to uncompressed PCM and writes it into AIFF, Apple's Audio Interchange File Format, so it drops straight into Logic, Pro Tools, Final Cut or any editor that prefers PCM over a compressed stream. The picture is discarded. Because AIFF is uncompressed, this page has no bitrate, quality or file-size controls at all — the size is pure arithmetic from sample rate, bit depth and channel count.

How to Convert MPEG-2 to AIFF

  1. Upload Your MPEG-2 File: Drag and drop your .mpeg2 file onto the page or click "+ Add Files". Several files can be queued and extracted in one batch.
  2. Choose the Audio Codec: PCM 16-bit Big Endian is the default and is what produces a genuine AIFF file. PCM 16-bit Little Endian, PCM A-law and PCM mu-law are also offered and produce AIFF-C variants instead — see the table below.
  3. Set Audio Sample Rate and Audio Channel: Leave Audio Sample Rate on ORIGINAL to keep the source rate (48000 HZ on DVD and broadcast material), or pick 48000, 44100, 24000, 16000, 12000 or 8000 HZ. Audio Channel takes Original, Stereo or Mono.
  4. Convert and Download: Use Trim to take a Start time and Duration if you only want one segment, then click Convert. Files upload over an encrypted connection, process on our servers, and are deleted automatically a few hours later.

AIFF or AIFF-C: What Each Audio Codec Option Writes

Apple published AIFF in 1989 and the extended AIFF-C in 1991. Both start with a FORM chunk, but the four bytes at offset 8 differ: a plain AIFF file reads AIFF, while a compressed or byte-swapped file reads AIFC. That single difference is what older Mac software checks, so pick deliberately.

Audio Codec option Container written Sample data Size vs 16-bit PCM Use when
PCM 16-bit Big Endian (default) FORM....AIFF 16-bit, big-endian — the original AIFF convention Baseline You want an unambiguous, universally readable AIFF
PCM 16-bit Little Endian FORM....AIFC 16-bit, little-endian (the sowt variant, headers stay big-endian) Same A tool specifically expects sowt-flavoured AIFF-C
PCM A-law FORM....AIFC 8-bit companded About half Telephony-grade speech archives
PCM mu-law FORM....AIFC 8-bit companded About half North American telephony material

You can verify which you got without any tooling: open the file in a hex viewer and read bytes 8 to 11 — AIFF or AIFC.

How Large the AIFF Will Be

Uncompressed PCM has a fixed cost per second: sample rate x bit depth x channels. AIFF also inherits IFF's 32-bit chunk sizes, so files much beyond 2 GiB become unreliable — worth knowing before you extract a three-hour broadcast capture in one pass.

Sample rate / channels Bytes per second Per minute Hours until ~2 GiB
48 kHz, 16-bit, stereo (typical DVD) 192,000 ~11.5 MB ~3.1
44.1 kHz, 16-bit, stereo 176,400 ~10.6 MB ~3.4
48 kHz, 16-bit, mono 96,000 ~5.8 MB ~6.2
16 kHz, 16-bit, mono (speech) 32,000 ~1.9 MB ~18.6
48 kHz, 16-bit, 5.1 (6 channels) 576,000 ~34.6 MB ~1.0

For comparison, 48 kHz 16-bit stereo PCM is 1,536 kbps, while the MP2 or AC-3 stream inside the MPEG-2 file was likely running at 192-448 kbps — so the AIFF is roughly 3 to 8 times larger without holding any more detail.

Common Errors and How to Fix Them

  • "The AIFF is enormous" — That is uncompressed PCM working as designed. Use Mono, a lower Audio Sample Rate, or Trim to the part you need. For a small shareable file, MPEG-2 to MP3 is the right tool; for lossless at roughly half the size, MPEG-2 to FLAC.
  • "It still sounds compressed" — PCM is a faithful decode of whatever was in the source. Artefacts from a 192 kbps MP2 track survive the transfer; AIFF stops further generational loss but cannot undo the first one.
  • "A long capture produced a broken or truncated file" — You are likely at the 2 GiB boundary. Extract in sections with Trim, or drop to Mono or a lower sample rate.
  • "My file is named .mpg or .vob" — This page's picker takes .mpeg2. They are the same program-stream family, but use the page that matches your extension: MPG to AIFF or VOB to AIFF.
  • "There is no bitrate setting" — Correct, and there should not be: PCM has no bitrate parameter to set. Rate, depth and channels are the only levers, and they are all on this page.

Frequently Asked Questions

Why does this page have no quality, bitrate or file-size options?

Because AIFF stores uncompressed PCM. There is no encoder making quality decisions, so there is nothing to tune — the entire File Compression section that appears on lossy audio targets is absent here by design. Your only size levers are Audio Sample Rate, Audio Channel, and how much of the file you keep with Trim.

Which option gives me a real AIFF rather than AIFF-C?

The default, PCM 16-bit Big Endian. It writes a FORM....AIFF header — the classic big-endian layout Apple defined in 1989 and the one every AIFF-aware application reads. The little-endian option produces the AIFF-C sowt variant, and the A-law and mu-law options produce AIFF-C as well, because companded data is by definition a compression type that plain AIFF does not describe.

Will AIFF restore quality the MPEG-2 audio lost?

No, and the size increase can make it look as if it did. MP2 and AC-3 are lossy; decoding them to PCM reproduces exactly what the decoder outputs and nothing more. A "lossless container" is not the same as "lossless audio". What AIFF buys you is a stable editing format that will not add another generation of loss every time you save.

What sample rate should I pick?

Leave it on ORIGINAL. DVD-Video and DVB/ATSC broadcast audio is 48000 Hz, so ORIGINAL keeps the samples untouched and avoids a needless resample. Choose a lower rate only when you deliberately want a smaller file for speech — 16000 HZ mono is about one sixth the size of 48 kHz stereo and is perfectly adequate for dialogue reference.

What happens to a 5.1 Dolby Digital soundtrack?

With Audio Channel on Original, the channel layout is carried through rather than folded down — AIFF's header describes the channel count, and the PCM encoders used here accept up to 8 channels, so a 5.1 AC-3 track decodes to six-channel PCM. Choose Stereo explicitly if the tool you are feeding wants two channels, and note the size table above: 5.1 at 48 kHz costs about 34.6 MB per minute.

Is there a maximum length for the AIFF?

Effectively yes. AIFF is built on IFF chunks whose sizes are 32-bit, which puts a practical ceiling a little over 2 GiB — roughly 3.1 hours at 48 kHz 16-bit stereo, or about an hour of 5.1. For anything longer, extract in sections using Trim, or convert to FLAC, which stays lossless without the ceiling.

AIFF or WAV — does it matter which I pick?

Not for audio quality; both hold identical PCM samples. The difference is byte order and ecosystem convention: AIFF is big-endian and the traditional Apple format, WAV is little-endian and the Windows one. Modern DAWs read both. If your workflow is Windows-first, use MPEG-2 to WAV instead and avoid explaining byte order to anyone.

Can I extract only one scene or song?

Yes — set Trim with a Start time and a Duration, in plain seconds or HH:MM:SS.sss, and only that span is decoded. Do it here rather than afterwards: trimming before extraction saves both the file size and the wait. The standalone audio trimmer does not accept video files, so trim on this page or cut the source first with the Video Cutter.

How is my file handled after upload?

It travels over an encrypted connection, is processed on our servers, and is deleted automatically a few hours after conversion — never shared, never made public, and no account is needed. The realistic limit on a large DVD-era capture is upload time, not the extraction. Once you have edited the AIFF, AIFF to FLAC will compress it losslessly for storage.

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