Convert MPEG to AIFC — Free Online Tool
Convert MPEG video files to AIFC audio, extracting the MP2 or AAC audio track and re-encoding it as big-endian PCM (pcm_s16be) — the same lossless PCM format used in professional Apple audio workflows. Ideal for recovering clean audio from legacy broadcast or DVD-compatible MPEG sources.
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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 contain a video stream (MPEG-1 or MPEG-2) alongside a compressed audio track — most commonly MP2, though AAC and MP3 are also possible. AIFC does not support video, so the video stream is discarded entirely during this conversion. The audio track is decoded from its lossy compressed form (MP2 or AAC) and re-encoded into pcm_s16be — 16-bit signed big-endian PCM — which is the default uncompressed audio codec for AIFC. This means the output is lossless PCM, but since the source audio was already lossy, the original compression artifacts are preserved in the final file. The result is a larger, uncompressed audio file with no further generational quality loss beyond what was already present in the MPEG source.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser version, this runs via FFmpeg.wasm compiled to WebAssembly — no installation needed and no files leave your device. |
-i input.mpeg
|
Specifies the input MPEG file, which may contain MPEG-1 or MPEG-2 video alongside a compressed audio track (typically MP2, MP3, or AAC). FFmpeg demuxes the container to access both streams. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM — the standard uncompressed audio encoding for AIFC files. The MP2 or AAC audio from the MPEG source is fully decoded and written as raw PCM samples, halting any further lossy compression. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. For pcm_s16be this parameter is effectively ignored because uncompressed PCM has a fixed bitrate determined by sample rate and bit depth, not a configurable target — it is included for consistency with other tool configurations. |
output.aifc
|
Defines the output filename with the .aifc extension, which tells FFmpeg to use the AIFC muxer. The video stream from the MPEG input is automatically dropped because AIFC is an audio-only container format. |
Common Use Cases
- Extracting the audio commentary or narration track from a legacy MPEG broadcast recording for use in an Apple Logic Pro or Pro Tools session as a PCM reference file
- Recovering the MP2 audio from MPEG-1 video files created by early DVD authoring tools and converting it to AIFC for archival in a professional audio library
- Pulling the audio from an MPEG-2 broadcast capture (e.g., a recorded TV segment) into AIFC format for import into Final Cut Pro as a clean audio-only clip
- Converting MPEG training or instructional videos from legacy CD-ROM software into AIFC so the audio can be edited and re-used in modern macOS audio tools
- Preparing audio extracted from DVD-compatible MPEG sources for delivery to a mastering engineer who requires big-endian PCM files in AIFC format
- Archiving the audio track of historical MPEG video recordings into an uncompressed AIFC format to prevent further lossy transcoding in future editing workflows
Frequently Asked Questions
No — the audio quality cannot improve beyond what was captured in the original MPEG file. The MP2 or AAC audio in the MPEG source is already lossy-compressed, so any artifacts from that compression are baked in. Converting to pcm_s16be in AIFC simply decompresses the audio into an uncompressed container, which stops any further quality degradation from additional re-encoding in future steps.
MPEG stores audio in a compressed format like MP2, which can be 10–20x smaller than uncompressed PCM. When the audio is decoded into pcm_s16be for the AIFC output, there is no compression applied — every sample is stored as a raw 16-bit value. For a 60-minute MPEG file, the extracted AIFC audio alone could be several hundred megabytes. If file size is a concern, consider using pcm_alaw or pcm_mulaw codecs, which apply light compression.
MPEG files support very limited metadata to begin with, and AIFC has its own minimal metadata structure. FFmpeg will attempt to map compatible metadata fields, but MPEG chapters and program-specific metadata are not preserved. AIFC also does not support subtitles, chapters, or multiple audio tracks, so only the primary audio stream and basic tags like title or artist may carry over.
AIFF (Audio Interchange File Format) stores only uncompressed PCM audio, while AIFC (AIFF-Compressed) is an extension that supports both uncompressed PCM codecs like pcm_s16be and compressed formats like pcm_alaw and pcm_mulaw. This tool outputs AIFC because it uses FFmpeg's AIFC muxer, which can accommodate all of these codec options under a single container format. For pcm_s16be specifically, the resulting file is functionally equivalent to AIFF in terms of audio data.
You can replace '-c:a pcm_s16be' with another supported AIFC codec to change the bit depth or encoding. For example, '-c:a pcm_s24be' gives you 24-bit PCM for higher dynamic range, and '-c:a pcm_f32be' uses 32-bit floating point. For a smaller file with light compression, try '-c:a pcm_alaw' or '-c:a pcm_mulaw'. Note that the '-b:a 128k' bitrate flag is ignored by uncompressed PCM codecs, so you can omit it when using pcm_s16be, pcm_s24be, or pcm_s32be.
Yes — on macOS or Linux, you can wrap the command in a shell loop: 'for f in *.mpeg; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.mpeg}.aifc"; done'. On Windows Command Prompt, use: 'for %f in (*.mpeg) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "%~nf.aifc"'. The browser-based tool processes one file at a time, but the FFmpeg command approach is well-suited for bulk processing large collections of legacy MPEG files.
Technical Notes
MPEG files use a program stream or transport stream container, and the audio codec is most commonly MP2 (MPEG-1 Audio Layer II) in legacy broadcast and DVD-compatible sources, though some MPEG-2 files carry AAC or MP3 audio. All of these are decoded during this conversion before being written as pcm_s16be in the AIFC container. The pcm_s16be codec uses 16-bit signed samples in big-endian byte order, which is the native byte order for AIFC (an Apple format designed on big-endian Motorola 68k and PowerPC hardware). Modern Apple software on ARM and Intel Macs reads this correctly without any byte-swapping overhead. The '-b:a 128k' flag in the command has no effect on pcm_s16be since uncompressed PCM has a fixed bitrate determined entirely by the sample rate and bit depth — for a 44.1 kHz stereo 16-bit file, the bitrate is always 1,411 kbps. If the source MPEG file contains multiple audio tracks or alternate language streams, FFmpeg will extract only the first (default) audio stream; to select a specific stream, add '-map 0:a:1' (or the desired stream index) to the command. AIFC does not support embedding subtitles, chapters, or video, so this conversion is strictly audio extraction.