Cut and trim AU (Sun Microsystems) audio files online. Extract segments from Unix and telephony recordings with compression control.
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00:00:05) and the length you want to keep (e.g. 00:00:30). Times are in hh:mm:ss format. Anything outside the window is discarded..au extension, same .snd magic header.The AU format was introduced by Sun Microsystems and became the de-facto sound format on early SunOS/Solaris workstations, NeXT computers, and the first wave of Java applets. Because Java 1.0/1.1's sun.audio.AudioPlayer and the early AudioClip API only accepted 8-bit μ-law AU at 8 kHz, a generation of Unix and Web audio assets was archived in this exact shape. Trimming AU directly — instead of round-tripping through MP3 or WAV — preserves the original encoding, sample rate, and the minimal 24-byte .snd header that legacy tooling expects.
.au clips can be trimmed without breaking the AudioClip decoder, which only handles a strict subset of AU encodings.Sound.h API expects AU; trimming at the source keeps the file directly usable in vintage software emulators (Previous, OPENSTEP/Mach)..snd magic number. Converting to WAV would require updating the consumer; trimming as AU does not.| Property | Value |
|---|---|
| Magic number | .snd (0x2E736E64), big-endian |
| Header size | 24 bytes (six 32-bit words) plus optional info chunk |
| Encodings | 8-bit μ-law, 8-bit A-law, linear PCM (8/16/24/32-bit), 32/64-bit IEEE float, ADPCM variants |
| Typical sample rates | 8000 Hz (telephony), 11025, 22050, 44100, 48000 Hz; some NeXT files use 8013 Hz |
| Channels | Mono (most legacy files) or stereo |
| Container | None — raw header-prefixed PCM/μ-law samples |
| Java Sound API support | Yes — read/write via AudioSystem since Java 1.2 |
| File extensions | .au, .snd |
| Property | AU | WAV | FLAC |
|---|---|---|---|
| Compression | μ-law / A-law / PCM | PCM / ADPCM | Lossless compression (~50%) |
| Header | 24 bytes, big-endian | 44+ bytes RIFF, little-endian | Variable, with metadata blocks |
| Best for | Telephony, legacy Unix/Java | Modern editing, broadcast | Archival music |
| Native streaming | Yes — header has known size | Trickier — chunk parsing | Yes |
| Metadata (tags) | Optional info chunk only | Limited (LIST INFO) | Rich (Vorbis comments) |
| File size at 8 kHz mono | ~28 MB/hr (μ-law) | ~28 MB/hr (μ-law in WAV) | ~14 MB/hr (PCM compressed) |
If your workflow is voice-over-IP, IVR, or any μ-law pipeline, stay in AU. If you need editor compatibility (Audacity, Pro Tools, Logic), convert with AU to WAV after trimming. For long-term archival of music-quality audio, AU to FLAC gives lossless compression and proper tagging.
| Preset / Mode | Typical bitrate | Use case |
|---|---|---|
| Quality Preset — Highest | ~256–320 kbps equivalent | Music-quality AU at 44.1/48 kHz |
| Quality Preset — High | ~192 kbps | Spoken-word podcasts, lectures |
| Quality Preset — Medium | ~128 kbps | General-purpose voice |
| Quality Preset — Low / Lowest | ~64–96 kbps | Voicemail, IVR prompts |
| Constant Bitrate — 64 kbps | 64 kbps | Telephony-grade μ-law (8 kHz mono) |
| Constant Bitrate — 32 kbps | 32 kbps | Compressed voice, dictation |
Because that is the historical default. Sun's original /dev/audio device on the SPARCstation 1 emitted exactly 8-bit μ-law at 8000 Hz, and Java 1.0/1.1's audio APIs were locked to that profile. Most AU files older than the early 2000s — voicemail dumps, classroom Java applets, museum exhibits — are still in that shape. Trimming preserves it; if you want higher fidelity, you'll need to re-record from the source rather than upsample.
Pick Quality Preset for music or general audio — the encoder allocates bits where they matter most and the file size scales with content. Pick Constant Bitrate when a downstream system expects a fixed kbps (telephony bridges, IVR providers, fixed-rate streaming). For 8 kHz μ-law telephony AU, Constant Bitrate at 64 kbps is the canonical setting.
They are sibling companding schemes defined in ITU-T G.711, but they use different logarithmic curves. μ-law is the North American and Japanese telephony standard; A-law is the European/ITU standard outside those regions. AU supports both, and the Java Sound API exposes them as AudioFormat.Encoding.ULAW and AudioFormat.Encoding.ALAW. If your file came from a North American carrier, it is almost certainly μ-law; from a European PBX, almost certainly A-law.
The xconvert pipeline re-encodes to keep the output bit-exact to your selected Quality Preset / sample rate / channel settings. If you specifically need a stream-copy trim (no re-encoding, instant cut), the FFmpeg command line equivalent is ffmpeg -ss <start> -t <duration> -i input.au -c copy out.au. The trade-off: a stream-copy cut will only land on encoder boundaries, which for μ-law is sample-accurate, but for ADPCM may snap to the nearest block.
No major modern browser ships a built-in AU decoder. Chrome, Firefox, Edge, and Safari rely on system codecs that historically supported AU on macOS (via Core Audio) but never on Windows or Linux without extra software. To preview an AU clip in a browser, convert it first with AU to MP3 or AU to OGG. For local playback, VLC, Audacity, and afplay (macOS) all decode AU natively.
Some early NeXT recordings used 8012.821 Hz (rounded to 8013) because the hardware clock divided differently from Sun's exact 8000 Hz. xconvert reads the header rate verbatim and re-samples to whatever you select in Audio Sample Rate. Picking 8000 Hz performs a tiny resample — inaudible for voice. Picking 22050 or 44100 Hz upsamples but does not add new high-frequency content; the original is band-limited to roughly 4 kHz.
No. Output retains the standard .snd (0x2E736E64) magic header in big-endian form, with the data offset, size, encoding, sample rate, and channel-count fields rewritten to match the trimmed segment. Any parser that handles standard Sun/NeXT AU — including Java's AudioSystem.getAudioInputStream, libsndfile, and FFmpeg's au demuxer — will read the result.
Yes — drag every file into the upload area and the same Trim window applies to each. This is the typical workflow for cleaning up a directory of .au voicemails (stripping the leading "you have a new message" prompt at a fixed offset). For per-file trim points, run them as separate jobs.
The trim-AU page is pre-configured for AU output: the format is locked, the encoding options match what AU supports (μ-law, A-law, PCM), and you avoid an accidental WAV/MP3 export. The general Audio Cutter accepts any input but defaults to a different output format. Use this page when the consumer of the clip explicitly requires .au; use the generic cutter when you want flexibility.