Cut WAV audio by setting start time and duration. No re-encoding preserves original lossless quality with fast processing.
Process files in seconds with our optimized servers
Frame-accurate cuts with intuitive timeline controls
Maintain original quality with smart re-encoding
12.5) and HH:MM:SS.sss (e.g. 00:00:12.500) for millisecond precision.WAV stores audio as raw uncompressed PCM samples wrapped in a RIFF container (introduced by IBM and Microsoft in 1991). Because every sample is independent and there are no inter-frame dependencies, a WAV file can be cut at any sample boundary, and rewriting PCM loses nothing — the audio you keep decodes to the same samples as the original. This makes WAV cutting the right step whenever quality, frame accuracy, or downstream re-editing matter.
/Ringtones.| Property | WAV (PCM) | FLAC | MP3 | AAC / M4A |
|---|---|---|---|---|
| Compression | None (raw PCM) | Lossless | Lossy | Lossy |
| Bitrate (CD-quality stereo) | ~1,411 kbps | ~700-900 kbps | 128-320 kbps | 96-256 kbps |
| Quality after a cut | Unchanged (PCM out) | Unchanged (lossless re-encode) | One lossy generation (near source bitrate) | One lossy generation (near source bitrate) |
| Generation loss on re-export | None | None | Slight (re-encoded) | Slight (re-encoded) |
| Classic file size cap | ~4 GB (32-bit RIFF size field) |
None (64-bit) | None practically | None practically |
| Best for | Editing, mastering, archiving | Lossless distribution | Streaming, podcasts | Apple ecosystem, streaming |
If you need true sample-accurate cuts and intend to keep editing afterwards, stay in WAV. Convert to MP3, FLAC, or M4A only as the final delivery step.
| Input | Example | Resolution | When to use |
|---|---|---|---|
| Seconds (decimal) | 12.5, 87.125 |
Down to 1 ms | Quick edits, short clips |
| HH:MM:SS.sss | 00:01:27.125 |
1 ms (3 decimal places) | Long files, podcast/interview timestamps |
At 44.1 kHz, 1 ms is roughly 44 samples; at 48 kHz it is 48 samples. That is well below the audible threshold for cut-point clicks, so HH:MM:SS.sss is precise enough for almost any practical edit.
No. WAV is uncompressed linear PCM, and the cut segment is written back out as PCM — the kept samples are unchanged, and nothing about the bit depth, sample rate, or channel layout changes. (Compare this with MP3 or AAC, where a cut re-encodes the audio and introduces one small generational loss.)
A classic WAV uses a 32-bit unsigned size field in its RIFF header, so files cap at just under 4 GiB — about 6.8 hours of 44.1 kHz / 16-bit stereo. Anything longer is typically stored as RF64 (specified by the EBU as Tech 3306, originally 2007, current Version 2 from June 2018), which extends the header to 64 bits and raises the ceiling to roughly 16 exabytes. Most professional field recorders (Sound Devices, Zaxcom, Sonosax) write RF64 automatically when a take crosses the 4 GB threshold.
Yes. Switch the Start time or Duration dropdown from Seconds to HH:MM:SS.sss. The .sss is milliseconds (three decimal places), which gives you a cut point accurate to about 44 samples at 44.1 kHz or 48 samples at 48 kHz — far below the threshold of audible clicks.
Yes. The output stays uncompressed PCM: bit depth (8, 16, 24, or 32-bit float), sample rate (8 kHz through 192 kHz and beyond), and channel count (mono, stereo, 5.1, multi-track) all pass through unchanged. The output file is the same PCM format as the input, just shorter.
Yes — both are WAV-compatible. BWF (Broadcast Wave Format) adds a bext metadata chunk for timecode, originator, and project info; RF64 swaps the RIFF 4-byte header for RF64 with a 64-bit size table to break the 4 GB barrier. The cut tool reads the audio data the same way for all three. Note that some BWF-specific metadata (timecode origin, take number) may not survive the cut — confirm in your DAW if you need it.
Upload the same WAV multiple times in the same session — once for each segment — and set a different Start time / Duration for each. Each pass produces an independent WAV. If you need to glue several non-contiguous pieces together afterwards, take the cut WAVs into Audacity or a DAW and concatenate them on a single track.
That click is a discontinuity: the cut happened in the middle of a non-zero waveform, so the output starts or ends mid-cycle. To avoid it, set your cut point on a zero-crossing (a moment of silence between syllables or notes), or apply a 5-20 ms fade-in/fade-out in a DAW after cutting. The cutting tool itself does not apply fades — it extracts exactly the range you specify.
Cut first. Cutting a WAV is lossless, so you keep all the headroom you need for the eventual compression. If you convert to MP3/AAC first and then cut, you have already taken the lossy hit, and cutting the compressed file adds a second lossy generation. For a workflow target like MP3 or M4A, cut here and convert afterwards.
Reach for Audio Cutter for a general format-agnostic cutter, Cut MP3 or Cut FLAC if your source is already compressed, and Compress WAV when the goal is shrinking the kept segment rather than trimming it.