Extract Audio from RMVB to AU — Free Online Tool
Extract audio from RMVB files and save it as AU format — Sun's classic Unix audio container using uncompressed PCM encoding. This tool decodes the AAC or MP3 audio stream from your RealMedia variable bitrate file and re-encodes it to 16-bit big-endian PCM, producing a lossless-quality AU file compatible with Unix systems and legacy audio tools.
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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
RMVB files store audio using lossy codecs — typically AAC or MP3 — wrapped in RealNetworks' variable bitrate container. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio to raw PCM samples. Those samples are then written into an AU file using the pcm_s16be codec (signed 16-bit, big-endian byte order), which is the standard encoding for Sun AU. Because the original RMVB audio is lossy, the output AU file reflects the quality ceiling of the source — the PCM encoding itself introduces no additional quality loss, but any compression artifacts from the original AAC or MP3 stream will be preserved in the output.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion. In the browser tool, this runs as FFmpeg.wasm compiled to WebAssembly, executing entirely on your device without any server upload. |
-i input.rmvb
|
Specifies the input file in RMVB format — a RealNetworks variable bitrate container that typically holds H.264 video alongside AAC or MP3 audio. FFmpeg reads both streams from this file before processing begins. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the video stream from the RMVB file. Since the goal is audio extraction, this prevents FFmpeg from attempting to encode or pass through the video track into the AU output, which is an audio-only format. |
-c:a pcm_s16be
|
Sets the audio codec for the output to pcm_s16be — signed 16-bit big-endian PCM — which is the standard and most compatible encoding for Sun AU files. This decodes the compressed AAC or MP3 audio from the RMVB and re-encodes it as uncompressed samples in the byte order expected by Unix AU players. |
output.au
|
Specifies the output filename with the .au extension, which tells FFmpeg to write a Sun AU container. The simple AU header encodes the sample rate, channel count, and pcm_s16be encoding type so any AU-compatible player or Unix audio tool can read it without additional configuration. |
Common Use Cases
- Importing audio from RMVB video files into Unix-based audio workstations or legacy Sun/NeXT software that only accepts AU format
- Extracting dialogue or music from old RealMedia video downloads to use in projects requiring uncompressed PCM audio
- Preparing audio extracted from RMVB content for analysis with signal-processing tools or scientific software that reads AU files natively
- Archiving the audio track from RMVB video files in an uncompressed format before the RMVB container becomes unplayable on modern systems
- Providing audio samples to legacy Unix systems or network appliances that use AU as their native audio format for playback or telephony
Frequently Asked Questions
The conversion involves one decode step — the AAC or MP3 audio inside the RMVB is decompressed to raw PCM — and then written into the AU container without any further lossy compression. This means no additional quality loss is introduced by the conversion itself. However, because the source audio in RMVB is already lossy, the AU output will reflect whatever compression artifacts existed in the original file — you cannot recover quality that was lost when the RMVB was first created.
RMVB stores audio using lossy compression (AAC or MP3), which dramatically reduces file size. AU with pcm_s16be stores every audio sample as a raw 16-bit integer with no compression whatsoever. A file that was compressed to 128 kbps AAC inside the RMVB will expand to approximately 1411 kbps as 16-bit stereo PCM at 44.1 kHz — roughly an 11x size increase. This is expected behavior for any conversion from compressed audio to uncompressed PCM.
AU files support several PCM encodings including pcm_s16be, pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw. The pcm_s16be (signed 16-bit big-endian) codec is chosen as the default because it offers the widest dynamic range among PCM options and is the most universally compatible format for AU files on Unix systems. The big-endian byte order reflects AU's origins on Sun SPARC hardware, and most AU-compatible software expects this format.
Yes — if you run the FFmpeg command locally, you can replace pcm_s16be with another supported AU codec. For telephony or VoIP applications, pcm_mulaw or pcm_alaw are common choices and reduce file size significantly while maintaining intelligibility for voice content. For example, use '-c:a pcm_mulaw' in place of '-c:a pcm_s16be' in the command. For general audio, pcm_s16be is recommended as it preserves the most dynamic range.
The Sun AU format has an extremely minimal header — it stores only sample rate, channel count, encoding type, and an optional free-form annotation field. It does not support structured metadata like title, artist, album, or cover art. Any metadata present in the RMVB file (such as RealMedia title tags) will not be carried over to the AU file, as the format simply has no mechanism to store it.
To change the output codec, replace 'pcm_s16be' with another supported AU codec like 'pcm_mulaw' or 'pcm_alaw'. To batch convert multiple RMVB files on the command line in a Unix shell, you can use: 'for f in *.rmvb; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.rmvb}.au"; done'. On Windows, use a for loop in PowerShell or Command Prompt. The browser tool processes one file at a time, so the FFmpeg command is the best option for bulk conversions or files larger than 1GB.
Technical Notes
RMVB (RealMedia Variable Bitrate) is a proprietary container that wraps video and audio using codecs common in early 2000s internet video distribution. The audio track is most commonly AAC or MP3 at bitrates ranging from 64 kbps to 192 kbps. When extracting to AU, FFmpeg fully decodes the compressed audio stream to PCM before writing it, so the output is always uncompressed regardless of the source bitrate. The AU format uses a simple fixed header (magic bytes 0x2E736E64) with no container overhead beyond basic stream parameters, making it trivial to parse but incapable of storing multi-track audio, chapters, or subtitles — none of which RMVB supports in this tooling context anyway. Sample rate is preserved from the source (typically 44100 Hz or 48000 Hz), and channel layout (mono/stereo) is carried through unchanged. The big-endian byte order of pcm_s16be is a firm characteristic of the AU format and cannot be changed without switching to a different container. If interoperability with modern DAWs or editors is the goal, WAV (pcm_s16le, little-endian) would be a more practical alternative, but AU remains the correct choice for Unix legacy compatibility.