Convert RMVB to AU — Free Online Tool
Convert RMVB video files to AU audio format, extracting the audio stream and encoding it as uncompressed 16-bit big-endian PCM — the native codec for Sun's AU container. This is ideal for Unix/Linux workflows that require raw, uncompressed audio from legacy RealMedia video content.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
RMVB files store video encoded with RealVideo and audio typically encoded with AAC or MP3 inside RealNetworks' variable-bitrate container. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio (AAC or MP3) from the RMVB file, then re-encodes it as uncompressed PCM signed 16-bit big-endian audio (pcm_s16be) inside an AU container. Because the audio must be fully decoded and re-encoded from a lossy compressed format into uncompressed PCM, this is a lossy-to-lossless pipeline in terms of the output format — however, the fidelity ceiling is limited by whatever quality was present in the original RMVB audio. The AU format uses a simple fixed header followed by raw audio samples, which makes the resulting file straightforward to parse on Unix systems but larger than the original compressed audio.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all demuxing, decoding, encoding, and muxing steps needed to convert the RMVB container and its compressed audio into the AU format. |
-i input.rmvb
|
Specifies the input file as an RMVB (RealMedia Variable Bitrate) file. FFmpeg reads the RealMedia container, identifies the video and audio streams (typically RealVideo and AAC/MP3), and prepares them for processing. |
-c:a pcm_s16be
|
Instructs FFmpeg to encode the output audio as signed 16-bit big-endian PCM, which is the standard and most compatible codec for the AU container format. This fully decodes the compressed AAC or MP3 audio from the RMVB and re-encodes it as raw, uncompressed audio samples in big-endian byte order as required by the Sun AU specification. |
output.au
|
Defines the output filename with the .au extension, which tells FFmpeg to wrap the pcm_s16be audio stream in a Sun AU container with the appropriate minimal header containing sample rate, bit depth, and channel count metadata. |
Common Use Cases
- Extracting dialogue or narration from RMVB lecture recordings or tutorial videos for ingestion into Unix-based audio processing pipelines that expect raw PCM audio
- Converting RMVB-distributed foreign-language films' audio tracks into AU format for use with legacy Sun Microsystems or Unix workstation audio tools that natively support the AU container
- Preparing audio from RMVB video content for analysis with scientific or signal-processing software on Linux that reads AU/PCM files directly without needing additional decoders
- Archiving the audio track from old RMVB-encoded internet video downloads into an uncompressed, container-simple format for long-term preservation on Unix file systems
- Feeding audio from RMVB video clips into older Java-based applications, which have historically supported AU (sun.audio) as a built-in audio format without external libraries
- Stripping and converting audio from RMVB screencasts or demos for playback testing on embedded Unix or Solaris systems with limited codec support
Frequently Asked Questions
No. While the output AU file uses uncompressed PCM audio (pcm_s16be), the fidelity is bounded by the original RMVB audio, which was already compressed with a lossy codec like AAC or MP3. The conversion decodes that lossy audio and stores the decoded samples as raw PCM — the artifacts and frequency limitations introduced during original RMVB encoding are preserved in the output. You get the convenience and compatibility of uncompressed PCM, but not higher quality than what was in the source.
The RMVB container stores audio as compressed AAC or MP3, which can achieve compression ratios of 10:1 or more compared to raw audio. The AU file stores every audio sample as an uncompressed 16-bit big-endian integer, so a one-minute audio track at 44100 Hz stereo requires roughly 10 MB with no compression at all. The RMVB file also contained a video stream, but that is discarded — the size difference you observe is almost entirely due to the shift from lossy compressed audio to uncompressed PCM.
The AU format has an extremely minimal header — it stores only technical parameters like sample rate, bit depth, channel count, and an optional free-form annotation field, but it has no standardized support for metadata tags like title, artist, or album. Any RealMedia metadata embedded in the RMVB source (such as title or author set during encoding) will be lost during this conversion. If metadata preservation is important, consider an intermediate format like FLAC or WAV, which support richer metadata fields.
The AU format technically supports several encodings including pcm_s16be, pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw. This tool defaults to pcm_s16be (signed 16-bit big-endian PCM) because it offers the best balance of fidelity and broad compatibility — it is the most universally recognised AU encoding and closely mirrors standard CD-quality audio bit depth. The big-endian byte order reflects the format's Sun SPARC workstation origins, where big-endian architecture was standard.
You can add the '-ar' flag to specify the sample rate and '-ac' to set the number of channels. For example, 'ffmpeg -i input.rmvb -c:a pcm_s16be -ar 44100 -ac 2 output.au' forces 44100 Hz stereo output. Without these flags, FFmpeg preserves the sample rate and channel count from the source RMVB audio, which is usually the right choice unless the destination system has specific requirements.
Yes. On Linux or macOS you can use a shell loop: 'for f in *.rmvb; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.rmvb}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.rmvb) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. The browser-based tool on this page processes one file at a time, so the FFmpeg command approach is especially practical when dealing with large collections of RMVB files.
Technical Notes
The AU format was developed by Sun Microsystems and became a staple on Unix and early Java platforms due to its simple fixed-size or variable-size header followed by raw audio data — there is virtually no framing or indexing overhead, which makes it easy to stream and parse programmatically. The pcm_s16be encoding uses big-endian byte ordering, which differs from the little-endian PCM used in WAV files; tools that read raw PCM must account for this byte order difference. Because RMVB audio is typically AAC at 128 kbps or similar, the decoded output will reflect that quality ceiling — the resulting AU file will be bit-for-bit reproducible from the same input but will not contain audio information that the original lossy codec discarded. The conversion strips the video entirely, so no video-related metadata, chapter markers, or subtitle tracks (none of which AU supports) are preserved. The AU format also does not support multiple audio channels beyond what fits in a single interleaved stream, and it has no native mechanism for embedded album art or extended tags. Files produced are fully compatible with Java's legacy 'sun.audio' package and standard Unix audio utilities like 'auplay' and 'sox'.