Convert FLV to AU — Free Online Tool
Convert FLV video files to AU audio format, extracting and re-encoding the audio stream as 16-bit big-endian PCM — the default codec for Sun's AU container. This is ideal for stripping Flash Video down to its raw, uncompressed audio for use on Unix/Linux systems or legacy audio pipelines.
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FFmpeg Command
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DownloadHow It Works
FLV files typically contain AAC or MP3 audio alongside an H.264 or Sorenson Spark video stream. During this conversion, FFmpeg discards the video entirely and decodes the compressed audio (AAC or MP3) from the FLV container, then re-encodes it as PCM signed 16-bit big-endian samples wrapped in Sun's AU container format. Unlike remuxing, this is a full transcode of the audio: the lossy compressed data is decoded to raw PCM, which means the output is uncompressed but cannot recover quality lost during the original FLV encoding. The AU format uses a minimal fixed header followed by raw PCM data, making the resulting file substantially larger than the FLV source.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary, the open-source multimedia processing engine that handles reading the FLV container, decoding its compressed audio, and writing the output AU file. |
-i input.flv
|
Specifies the input Flash Video file. FFmpeg reads the FLV container, identifying the video stream (typically H.264) and audio stream (typically AAC or MP3) packaged inside it. |
-c:a pcm_s16be
|
Sets the audio output codec to PCM signed 16-bit big-endian — the default and most compatible encoding for the AU format. This fully decodes the compressed AAC or MP3 audio from the FLV and re-encodes it as uncompressed 16-bit PCM samples in network byte order (big-endian), as expected by Sun AU readers. |
output.au
|
Defines the output filename with the .au extension, which tells FFmpeg to use the Sun AU muxer. The resulting file contains the AU magic number header followed by raw pcm_s16be audio data, with no video or metadata from the original FLV. |
Common Use Cases
- Feeding audio from Flash Video archives into Unix or legacy audio processing tools that natively read AU files, such as older SunOS or Solaris audio utilities
- Extracting uncompressed PCM audio from FLV recordings — such as lecture captures or old Flash-based screencasts — for ingestion into professional audio editing software that requires raw PCM input
- Archiving the audio track of historical FLV web content in an uncompressed format before the source FLV files become unplayable due to Flash Player deprecation
- Preparing audio from FLV streams for analysis or signal processing pipelines on Unix systems where AU is the expected input format
- Converting FLV-based podcast or interview recordings to AU so they can be played back on early internet audio infrastructure or emulated Unix environments
- Stripping and decompressing audio from FLV game captures or screencasts to use as raw waveform input for custom audio fingerprinting or transcription workflows
Frequently Asked Questions
There is an inherent quality ceiling imposed by the original FLV encoding. FLV stores audio as AAC or MP3, both of which are lossy formats, so some quality was already lost when the FLV was created. Converting to AU with pcm_s16be fully decodes that compressed audio to uncompressed PCM, so no additional lossy compression is applied — but the artifacts from the original AAC or MP3 encoding are baked in and cannot be recovered. The output AU file is a faithful, uncompressed representation of whatever audio quality existed in the FLV.
FLV stores audio using lossy compression (AAC or MP3), which can reduce file size by 5–10x or more compared to raw PCM. The AU format with pcm_s16be stores every audio sample as an uncompressed 16-bit value, so a 10-minute FLV with 128k AAC audio might produce an AU file that is 50–100 MB or larger. This is expected behavior — AU is a raw waveform format, not a compressed one.
Yes, the AU format supports multi-channel PCM audio including stereo. If your FLV contains a stereo audio track, FFmpeg will preserve both channels in the pcm_s16be output. The AU header encodes the number of channels, so a stereo FLV source will produce a stereo AU file with the correct channel count declared in the header.
You can add the '-ar' flag to specify a target sample rate. For example, 'ffmpeg -i input.flv -c:a pcm_s16be -ar 44100 output.au' will resample the audio to 44,100 Hz. If you omit '-ar', FFmpeg preserves the original sample rate from the FLV source, which is typically 44100 Hz or 48000 Hz for AAC audio.
Yes. The AU format supports several PCM variants. You can substitute '-c:a pcm_s16be' with '-c:a pcm_mulaw' for G.711 mu-law (common in telephony), '-c:a pcm_alaw' for A-law encoding, '-c:a pcm_s8' for 8-bit signed PCM, or '-c:a pcm_u8' for 8-bit unsigned PCM. The mu-law and A-law options produce smaller files than 16-bit PCM but with reduced dynamic range, while 8-bit options are the most compact but lowest fidelity.
Yes. On Linux or macOS you can use a shell loop: 'for f in *.flv; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.flv}.au"; done'. On Windows Command Prompt, use 'for %f in (*.flv) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. This applies the same pcm_s16be conversion to every FLV in the current directory and names each AU output after its corresponding source file.
Technical Notes
The Sun AU format was one of the earliest audio formats used on the internet and remains common in Unix/Linux legacy contexts. Its header is intentionally minimal: it encodes encoding type, sample rate, channel count, and an optional annotation field, then immediately precedes the raw audio data. Because AU with pcm_s16be uses big-endian byte ordering (network byte order), it is natively compatible with older SPARC and MIPS-based Unix workstations. When converting from FLV, the video stream is silently dropped — FFmpeg does not error on this, it simply has no video output codec mapped for AU. The sample rate from the FLV source is preserved by default; if the source AAC track is 48 kHz, the AU output will also be 48 kHz. Metadata such as title or artist tags stored in the FLV container are not carried over to the AU file, as AU's annotation field is not populated by FFmpeg's default muxer. There is no chapter or subtitle support in AU, and since the output is a single uncompressed stream, the resulting file is not suitable for streaming in the modern HTTP sense despite AU technically supporting sequential reads.