Convert SWF to AU — Free Online Tool
Extract and convert audio from SWF Flash files to Sun AU format using PCM 16-bit big-endian encoding. This tool strips the MP3 or AAC audio track from a Shockwave Flash file and re-encodes it as uncompressed PCM audio in the classic Unix AU container — ideal for legacy Unix systems and audio analysis workflows.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
SWF files typically carry audio encoded as MP3 (via the libmp3lame codec) or AAC. During this conversion, FFmpeg demuxes the SWF container to extract the compressed audio stream, then fully decodes and re-encodes it as PCM signed 16-bit big-endian audio (pcm_s16be) wrapped in the Sun AU container. The video, vector graphics, and interactive elements embedded in the SWF are discarded entirely — AU is a pure audio format with no video support. Because the SWF audio is lossy (MP3 or AAC) and the output is uncompressed PCM, the conversion produces a lossless representation of the already-compressed source audio; no additional lossy degradation occurs, but the original compression artifacts from the SWF's audio encoding remain.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all demuxing, decoding, re-encoding, and muxing steps needed to convert the SWF file to AU format entirely within a single process. |
-i input.swf
|
Specifies the input SWF file. FFmpeg will parse the Shockwave Flash container to locate the embedded audio stream (typically MP3 via libmp3lame or AAC), which will then be decoded for re-encoding into PCM. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, the default and most common encoding for Sun AU files. This produces uncompressed audio data stored as 16-bit samples in big-endian byte order, matching the original Sun SPARC architecture convention for AU files. |
output.au
|
Specifies the output filename and triggers FFmpeg to use the Sun AU container format. The .au extension is recognized by FFmpeg as the Sun AU muxer, which writes the standard four-byte magic number, minimal header fields, and raw PCM sample data. |
Common Use Cases
- Recovering audio content from archived SWF Flash animations or games for use on Unix/Linux legacy systems that natively support the AU format
- Feeding SWF-sourced audio into Unix audio processing pipelines or older signal analysis tools that expect raw PCM in AU containers
- Extracting voiceover or sound effects from old Flash-based e-learning modules to repurpose as uncompressed audio assets
- Converting SWF audio tracks to AU for playback on early Sun workstations or retro computing environments where AU was the standard audio format
- Preparing audio from Flash animations for ingestion into broadcast or archival systems that require uncompressed PCM rather than compressed codecs like MP3 or AAC
- Debugging or inspecting the raw audio content of a SWF file by converting it to a simple, headerless-style PCM format where the waveform data is straightforward to analyze
Frequently Asked Questions
Not in a meaningful way beyond what already exists. The audio inside a SWF file is typically compressed as MP3 or AAC, which introduced lossy artifacts when the SWF was originally created. This conversion fully decodes that compressed audio and stores it as uncompressed PCM in the AU file, so no new lossy compression is applied. The output is a perfect PCM representation of the already-compressed source — you cannot recover quality lost during the original SWF encoding, but you will not degrade it further.
All of it is discarded. The AU format is a purely audio container with no support for video streams, vector graphics, or interactivity. FFmpeg extracts only the audio track from the SWF during this conversion. If you need to preserve the visual content, you would need to first convert the SWF to a video format like MP4 before performing any audio extraction.
SWF files store audio in compressed formats like MP3 or AAC, which dramatically reduce file size. The AU output uses pcm_s16be — uncompressed 16-bit PCM audio — which stores every sample as raw data with no compression. A one-minute audio clip at 44.1kHz stereo will be approximately 10MB as PCM, compared to under 1MB as MP3 at 128kbps. This size increase is expected and is a characteristic of switching from lossy compressed audio to uncompressed PCM.
Replace 'pcm_s16be' in the '-c:a pcm_s16be' flag with another codec supported by the AU format. Valid options include 'pcm_s8' for 8-bit signed PCM, 'pcm_u8' for 8-bit unsigned PCM, 'pcm_alaw' for G.711 A-law companding (common in telephony), and 'pcm_mulaw' for G.711 mu-law companding. For example: 'ffmpeg -i input.swf -c:a pcm_mulaw output.au' would produce mu-law encoded AU audio, which is smaller than 16-bit PCM and widely used in Unix telephony applications.
Yes, on the command line you can use a shell loop to process multiple files. On Linux or macOS, run: 'for f in *.swf; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.swf}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.swf) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. Each SWF file will be processed individually and produce a corresponding AU file with the same base name.
Almost certainly not. SWF files do not carry standard audio metadata tags in the way that MP3 or MP4 containers do, and the AU format itself has only a minimal header with fields for data offset, data size, encoding type, sample rate, and channel count. There is no standard metadata tag structure in AU comparable to ID3 or Vorbis comments, so no title, artist, or description information will be embedded in the output file.
Technical Notes
The Sun AU format uses a straightforward binary header followed by raw audio sample data, making it one of the simplest audio containers in existence. The default codec for this conversion, pcm_s16be, stores audio as 16-bit signed integers in big-endian byte order — a reflection of the Sun SPARC architecture's big-endian nature. This byte order matters if you intend to read the raw sample data programmatically on a little-endian system (such as x86), as you will need to byte-swap the samples. SWF files can contain multiple audio streams tied to different timeline events or interactive triggers; FFmpeg will typically select the primary or first detected audio stream for extraction. If your SWF has audio at a non-standard sample rate, FFmpeg will preserve that rate in the AU header rather than resampling, so downstream tools must support the original rate. The AU format has no concept of chapters, subtitle tracks, or multiple audio streams, and has no DRM support — all characteristics that align well with its Unix legacy use case but limit its utility in modern consumer media workflows.