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 boundary by simply copying the bytes between two sample offsets and rewriting the RIFF and data chunk headers. That is the entire operation — the audio you keep is bit-identical to 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 |
| Frame-accurate cut without re-encode | Yes (per sample) | Yes (per FLAC frame) | No (MP3 frames are ~26 ms; cuts snap to frame boundary) | No (AAC has 1024-sample frames + encoder delay/padding) |
| 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, so a cut is just a byte-range copy — the kept samples are bit-identical to the source. There is no decode/encode step, so nothing about the bit depth, sample rate, or channel layout changes. (Compare this with MP3 or AAC, where any cut snaps to an encoder frame boundary and re-encoding introduces a 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. Because no codec is involved, the 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 is a pure sample-range copy.
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 the cut will additionally snap to the encoder's frame boundary (~26 ms for MP3, ~21 ms for AAC at 48 kHz). 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.