An online audio cutter and joiner that runs entirely in your browser trims one continuous time range from a decoded audio file and exports the result as an uncompressed PCM16 WAV — the source file is never uploaded to a server. People searching for "audio cutter and joiner online" usually want two related jobs done: clip a precise segment from a recording (a podcast intro, a song chorus, a voice memo) and, sometimes, combine several clips into one track. The Audio Cutter tool covers the first job end to end with no upload, no account, and no codec installation. You pick a file, wait while the browser's Web Audio decoder converts it to an AudioBuffer, type two timestamps in seconds, and save a freshly encoded WAV. For the second job — stitching multiple files in a chosen order — the companion Audio Joiner tool handles that second job. Knowing which tool does which keeps your workflow honest: cutting extracts a range from one file. The Audio Cutter works locally and produces a PCM16 WAV.
How the Browser Does the Cut Without a Server
The cut happens through the browser's Web Audio API, which is the same pipeline documented on the MDN decodeAudioData reference. When you select a file, the page reads it as a blob, checks the file name or MIME type, enforces a 25 MB byte cap, and calls decodeAudioData on an AudioContext. The browser may resample the source to the AudioContext's native rate during decoding, which means the displayed sample rate describes the decoded working buffer, not necessarily the encoded bitstream rate. The page shows the decoded duration, sample rate, channel count, and frame count before you cut, so the numbers you see are the numbers the exporter will use. Nothing leaves the tab; temporary object URLs are released when you replace a file, cut again, or close the page, and in-flight reads are guarded so an older job cannot overwrite a newer selection.
Limits, Formats, and What the Picker Recognizes
The picker accepts file names and MIME types for common MP3, WAV, M4A, AAC, Ogg, WebM, and FLAC files, but recognition is only a first filter. Whether the browser can actually decode the audio depends on the codecs your current Safari, Chrome, Firefox, or other browser ships with, and that support differs across operating systems and versions. An unsupported, damaged, mislabeled, encrypted, or incomplete file produces a clear decode error rather than a silent substitution. After decoding, the working buffer is checked against explicit safety limits rather than being truncated silently.
| Limit | Maximum |
|---|---|
| Encoded source file size | 25 MB |
| Decoded duration | 15 minutes |
| Channel count | 8 |
| Decoded sample rate | 192 kHz |
| Total channel-samples across all channels | 30,000,000 |
The channel-sample limit is the one that bites first for high-rate stereo or multichannel files. A 15-minute stereo track at 48 kHz would decode to 15 × 60 × 48,000 × 2 = 86,400,000 channel-samples, which is rejected well before the duration cap. Conversely, a one-minute mono clip at 22.05 kHz uses only about 1.3 million channel-samples, so small voice memos and notification sounds sail through. Browser decoding necessarily happens before the decoded duration and sample count can be inspected, so a highly compressed long file may still consume temporary memory during the decode attempt.
How to Cut an Exact Time Range in Your Browser
Open the Audio Cutter page and follow these steps:
- Choose a browser-decodable audio file within the displayed file and decoded-audio limits.
- Read the decoded duration, sample rate, channel count, and frame count shown on the page so you know exactly what the working buffer contains.
- Enter a non-negative start time in moments and a later end time that does not exceed the exact decoded duration. The page rounds both to the nearest decoded sample-frame boundary using the AudioBuffer sample rate, with the start frame included and the end frame excluded. A selection that rounds to less than one frame is rejected rather than producing an empty file.
- Select "Cut and export WAV" and wait for the exporter to finish. The production exporter and isolated tests share the same frame-range function, so the displayed cut duration equals the selected frame count divided by the decoded sample rate.
- Preview the resulting PCM16 WAV in the on-page player. Listen for the spoken word or musical transient at both boundaries; if either sounds clipped, adjust the timestamps and cut again — changing either time field clears the previous WAV, and cutting again releases the prior download URL.
- Click the download button to save a freshly encoded RIFF/WAVE file with interleaved, little-endian, signed 16-bit PCM samples. The page reports the output duration and file size before download.
Why the Output Is Always an Uncompressed WAV
Every download is a newly encoded PCM16 WAV. The exporter does not retain the source's MP3, AAC, Vorbis, Opus, FLAC, or other compressed encoding, and it does not preserve tags, album art, cue points, chapters, loudness metadata, encoder settings, or any other container fields. Floating-point Web Audio samples are clamped to the PCM range from -1 through 1, with negative full scale written as -32768 and positive full scale as 32767; non-finite values become silence. Multiple channels stay separate and are interleaved in their original order, up to the eight-channel limit. Because PCM WAV stores uncompressed samples, a cut can be substantially larger than the source. The data size follows a fixed formula, and the page shows both the duration and the file size before you click download.
| Field | Value |
|---|---|
| Selected length | 60.000 seconds |
| Decoded sample rate | 44,100 Hz |
| Channels | 2 (stereo) |
| Frames in selection | 60 × 44,100 = 2,646,000 |
| Bytes per sample | 2 (signed 16-bit) |
| Data size | 2,646,000 × 2 × 2 = 10,584,000 bytes |
| RIFF/WAVE header | 44 bytes |
| Total download size | 10,584,044 bytes ≈ 10.1 MB |
A 60-second MP3 encoded at 128 kbps holds about 60 × 16,000 = 960,000 bytes of audio data, so the same range exported as PCM16 WAV is roughly eleven times larger in raw bytes. Plan storage and upload bandwidth around the WAV size, or transcode the result to a compressed format afterwards if size matters more than fidelity.
Cutting vs Joining Audio Files
Cutting and joining are different operations and the page you choose should match the job. The Audio Cutter extracts one continuous range from a single decoded file and exports a WAV of exactly that range. If your goal is a single ringtone, voicemail greeting, or podcast intro, cutting from one source is the right path. If your goal is a playlist-style mashup or a stitched interview with several takes, joining is the right path. The Audio Cutter runs locally, produces PCM16 WAV, and does not upload your source.
Tips for a Clean, Verifiable Cut
Keep the original file alongside the WAV until you have verified the cut. Play the preview at both boundaries: a spoken word that begins with a hard consonant can sound clipped if the start frame lands mid-phoneme, and a drum hit at the end frame may lose its transient if the boundary cuts through the attack. Use the page's reported sample rate as the basis for any time math, because the decoded buffer may have been resampled to the AudioContext rate during decode — the timestamps are rounded at the decoded rate, not the encoded rate. If you need waveform-level fades, lossless compressed-frame cutting, or metadata preservation, move to a dedicated editor; the Audio Cutter is designed for one explicit job and refuses to silently substitute behaviour it doesn't support, so it does not offer a target resampling control, remix, normalize, fade, or compress the WAV.