Extract Audio from FLV to AU — Free Online Tool
Extract audio from FLV (Flash Video) files and save it as an AU file using PCM 16-bit big-endian encoding — a lossless, uncompressed format native to Unix systems. This tool strips the H.264 or FLV video stream and decodes the AAC or MP3 audio into raw PCM, giving you a clean, uncompressed AU file entirely in your browser.
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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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Settings
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
FLV files typically carry AAC or MP3 audio alongside a video stream encoded with H.264 or the older FLV codec. During this conversion, the video stream is completely discarded (not re-encoded — just dropped), and the compressed audio (AAC or MP3) is fully decoded and re-encoded into PCM 16-bit big-endian (pcm_s16be), the default codec for the AU container. This means the output is uncompressed PCM audio — every sample is stored as a raw 16-bit signed integer in big-endian byte order, exactly as Sun Microsystems defined the AU format. The result is a lossless representation of whatever audio was in the FLV, though any quality loss introduced by the original AAC or MP3 compression in the source file cannot be recovered.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In the browser-based tool, this runs via FFmpeg.wasm (WebAssembly) entirely client-side — your FLV file never leaves your machine. |
-i input.flv
|
Specifies the input Flash Video file. FFmpeg will detect the FLV container and identify the contained streams, typically an H.264 or FLV video stream and an AAC or MP3 audio stream. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the video stream in the FLV. This is essential for audio extraction — without it, FFmpeg would attempt to include video in the output, which the AU format cannot contain. |
-c:a pcm_s16be
|
Decodes the compressed AAC or MP3 audio from the FLV and re-encodes it as 16-bit signed PCM in big-endian byte order, which is the default and most common encoding for the AU container format originating from Sun Microsystems Unix systems. |
output.au
|
Sets the output filename with the .au extension. FFmpeg uses this extension to automatically select the Sun AU muxer, which writes the AU file header (magic number, sample rate, channel count, encoding type) followed by the raw PCM audio samples. |
Common Use Cases
- Recovering audio from legacy Flash-era video archives for use in Unix or Linux audio pipelines that expect AU-format input
- Feeding FLV-sourced audio into Sun/Oracle audio workstation tools or older Java-based media applications that natively support the AU format
- Decompressing AAC or MP3 audio from a Flash video file into uncompressed PCM for lossless editing or further processing in audio software
- Extracting clean audio from FLV screen recordings or webinar archives for use in academic or scientific data analysis tools that consume raw PCM AU files
- Converting Flash video audio to a byte-order-explicit PCM format (big-endian) for systems or embedded environments that require network byte order audio
- Archiving audio from old FLV-format video content into a simple, headerless-adjacent format that will remain readable without proprietary codecs
Frequently Asked Questions
No — the AU output will not be higher quality than the original FLV audio. FLV files store audio in compressed formats like AAC or MP3, both of which are lossy. When converted to AU, the compressed audio is decoded into uncompressed PCM, which accurately represents what the decoder produces, but cannot recover detail lost during the original AAC or MP3 encoding. The AU file will be lossless from this point forward, but it reflects the quality ceiling of the FLV's source audio.
The AU format stores audio as raw, uncompressed PCM samples — in this case, 16-bit signed big-endian integers at whatever sample rate the source audio uses. FLV files use AAC or MP3 compression, which typically reduces audio data by 90% or more. When you decompress that audio into PCM for the AU file, you get the full uncompressed size. For example, a 10-minute FLV with 128kbps AAC audio might produce an AU file of roughly 100MB or more, depending on the sample rate and channel count.
AU files have limited native support on modern desktop operating systems. macOS can open AU files via QuickTime Player, but Windows has no built-in AU playback. On both platforms, VLC Media Player and Audacity handle AU files reliably. The AU format was designed for Unix systems, so it is most natively supported in Linux/Unix environments, which is a key reason to choose this format for Unix-oriented audio workflows.
The AU format supports multiple channels, including stereo, and FFmpeg will preserve the channel layout from the FLV source. If the FLV contains stereo AAC or MP3 audio, the output AU file will also be stereo PCM. The channel count is encoded in the AU file header alongside the sample rate and encoding type, so compliant AU players will correctly interpret the channel configuration.
Replace '-c:a pcm_s16be' with your preferred AU-compatible codec. For mu-law encoding (common in telephony), use '-c:a pcm_mulaw'; for A-law, use '-c:a pcm_alaw'; for 8-bit signed PCM, use '-c:a pcm_s8'. For example: 'ffmpeg -i input.flv -vn -c:a pcm_mulaw output.au'. Note that mu-law and A-law are companding codecs that reduce file size at the cost of dynamic range, whereas pcm_s16be gives you the highest fidelity AU output.
Yes, you can adapt the command for batch processing in a shell script. On Linux or macOS, use: 'for f in *.flv; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.flv}.au"; done'. On Windows Command Prompt: 'for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.au"'. The browser-based tool processes one file at a time, but the displayed FFmpeg command is exactly what runs locally, making it straightforward to script for large collections.
Technical Notes
The AU format uses a minimal fixed header (24 bytes minimum) that stores the data offset, data size, encoding type, sample rate, and channel count — after which raw PCM samples follow immediately. Using pcm_s16be means each sample is a 16-bit signed integer stored in big-endian (network) byte order, which is the canonical byte order for AU and reflects its Sun Microsystems/SPARC origins. FFmpeg correctly writes the AU magic number (.snd) and header fields during this conversion. One important limitation: the AU format does not support metadata tags (no ID3, no Vorbis comments), so any title, artist, or album information embedded in the FLV's audio stream will be lost. The FLV container also does not support multiple audio tracks, so there is no ambiguity about which stream to extract. If your FLV source uses a 44.1kHz stereo AAC audio track (the most common configuration for Flash web video), the output AU file will be approximately 10MB per minute of audio at 16-bit stereo 44.1kHz, compared to roughly 1MB per minute for 128kbps AAC.