A cut audio result that looks wrong is almost always the symptom of one of three things: the start or end time was rounded to the nearest audio frame, the file was re-encoded as an uncompressed PCM16 WAV instead of staying in its original compressed format, or the times you entered exceeded the decoded duration of the source. The Audio Cutter trims one continuous range from a decoded audio buffer entirely inside your current browser tab. Because every download is a freshly written PCM16 WAV with no original tags, no album art, no cue points, and no source codec, the file you get back will not match the original file's size, extension behavior, or metadata. Re-cutting with verified times, a confirmed decoded duration, and an understanding of the new container gives you a clean result without uploading the original anywhere. The browser does the decode, the frame selection, and the WAV write locally, so fixing a wrong-looking cut means re-running those same steps with corrected inputs, not chasing a corrupted file.
Common Reasons a Cut Result Looks Wrong
The most common causes of a result that looks wrong are value problems and format expectations rather than tool failures. Each of these issues has a specific cause that the Audio Cutter's display can confirm before you cut again.
Boundary rounding at sample frames
Start and end seconds are converted to the nearest decoded sample-frame boundary using the AudioBuffer sample rate. The start frame is included and the end frame is excluded. If you enter 1.234567 seconds at a 44.1 kHz decoded rate, the tool snaps to the nearest frame, so the actual cut can be a fraction of a millisecond different from what you typed. For most content this difference is invisible, but at very short cuts or near musical transients the rounding can move the boundary enough to feel off. Re-cut using values that are clearly inside the content you want to keep, and preview both boundaries.
Format change from MP3 or AAC to WAV
Every result is a newly encoded RIFF/WAVE file containing interleaved, little-endian, signed 16-bit PCM samples. The original MP3, AAC, Vorbis, Opus, FLAC, or other compressed encoding is not retained. If you were expecting the file to stay small and lossy, a 5 MB MP3 can produce a 30 MB WAV at the same duration. That is not damage; it is the math of uncompressed audio: output data uses two bytes for every selected frame in every channel, plus a 44-byte header.
Times that exceeded the decoded duration
The page displays the decoded duration before you cut. End cannot exceed that exact duration, start cannot be negative, and end must be greater than start. If your end value was a half-second past the displayed duration, the tool rejects the selection rather than producing a clipped file. A rejected selection looks like nothing happened, which can read as a wrong result if you were expecting a download.
Codec or container not supported by your browser
Input support depends on your browser's Web Audio decoder. The file picker recognizes common MP3, WAV, M4A, AAC, Ogg, WebM, and FLAC names and MIME types, but recognition does not guarantee that every codec variation can be decoded by every browser. Container and codec support differs across Safari, Chrome, Firefox, operating systems, and browser versions. An unsupported, damaged, mislabeled, encrypted, or incomplete file produces a clear decode error rather than a silent empty result.
What the Audio Cutter Shows Before You Cut Again
Before you re-cut, read the values the page prints about your decoded buffer. These are the numbers that decide where the boundary actually lands and how the WAV will be written.
- Decoded duration in seconds. This is the exact maximum end value you can enter.
- Sample rate of the decoded buffer in Hz. Web Audio may resample the source to the AudioContext sample rate while decoding, so this describes the working buffer and the exported WAV, not necessarily the source file's encoded rate.
- Channel count. The number of channels the decoded buffer holds, up to eight. If the source is stereo, the result stays stereo; if it has more channels, those are retained up to the stated limit.
- Frame count. The total number of frames across all samples in the buffer.
- Reported output duration and file size. Shown before download so you can sanity-check the selection before committing.
Displayed cut duration is the selected frame count divided by the decoded sample rate (see the MDN AudioBuffer reference for the underlying sample-rate semantics), not an unrelated timer estimate. The production exporter and isolated tests share the same frame-range function, so the math you see on the page is the math that lands in the file.
How to Re-Cut the Audio in Your Browser
- Open the tool and pick the same source file again. Choosing another file clears the old decoded buffer, players, errors, and result, so you start from a known state. The file picker recognizes common MP3, WAV, M4A, AAC, Ogg, WebM, and FLAC names and MIME types.
- Wait for Web Audio to decode the file. The page will print the decoded duration, sample rate, channel count, and frame count. If decoding fails, the page shows a clear decode error rather than a silent empty buffer; that error usually means the codec variation is not supported by your browser, the file is damaged, or the file exceeds the 25 MB encoded limit.
- Note the decoded duration in moments. This is the longest value you can use for end. Use it as a reference for any future cut on the same source.
- Enter a non-negative start time and a later end time. Start cannot be negative or at the end of the audio. End must be greater than start and cannot exceed the decoded duration. A selection that rounds to less than one frame is rejected rather than producing an empty file.
- Select Cut and export WAV. The tool encodes the selected range as interleaved signed little-endian PCM16 with a 44-byte RIFF/WAVE header. Changing either time field clears the previous WAV, so a new export always replaces the prior download URL.
- Preview the result in the page's player. Listen for the spoken words or musical transients right at both boundaries. If a transient at the start is cut off, move start earlier; if the end clips a held note, move end later. Boundary snap is to the nearest decoded sample frame, so small moves in your time value do translate to small moves in the file.
- Check the reported output duration and file size before downloading. Size in bytes is the selected frame count multiplied by the channel count and two bytes per sample, plus a 44-byte WAV header.
- Download the WAV. The browser saves it locally. Nothing has been uploaded; reading, decoding, frame selection, encoding, preview, and download all happened inside your current tab.
Limits That Can Make a Result Look Broken
The Audio Cutter applies hard safety limits and surfaces them with explicit messages rather than silently trimming an over-limit file. If your result looks wrong, check whether any of these were hit.
| Limit | Maximum | Where it shows up |
|---|---|---|
| Encoded input file size | 25 MB | Rejected before decoding if the file on disk is larger |
| Decoded duration | 15 minutes | Rejected after decoding if the buffer is longer |
| Sample rate | 192 kHz | Rejected if the decoded buffer exceeds this rate |
| Channel count | 8 | Rejected if the decoded buffer exceeds this count; otherwise retained |
| Total samples across all channels | 30,000,000 | High-rate or multichannel files can hit this before the duration limit |
Browser decoding necessarily happens before decoded duration and sample count can be inspected, so a highly compressed long file may still require temporary memory during the decode attempt. An over-limit file is rejected with a message rather than truncated.
If your selected source is stereo, the result stays stereo; if it has more channels, those are retained up to the stated limit. Playback software must support the resulting channel layout, otherwise the file can appear silent or unplayable in a player that does not understand the layout. That is not a tool bug, but it can read as wrong when you expected stereo and got something else.
Check the New WAV Before You Trust It
The new PCM16 WAV is independent of the original file's compression, metadata, and channel count, so it deserves its own sanity check before you use it in a project. For a deeper walk-through of result verification, see how to check the result after you cut audio.
- Duration. Open the WAV in any player and confirm the length matches the reported output duration on the page.
- Channels. Confirm stereo stayed stereo, or that a multichannel source retained the expected channel count up to eight.
- Boundary content. Listen at both the start and end. If a spoken word is half-eaten or a drum hit is muted, re-cut with start or end shifted by a few hundred milliseconds and try again.
- File size. Confirm the size matches the page's reported output size. PCM16 WAV is uncompressed, so a short MP3 source can produce a noticeably larger WAV; that is expected, not corruption.
If any of these checks fail, the simplest recovery is to re-run the cut with adjusted times. In-flight reads and decodes are guarded so an older job cannot overwrite a newer selection, and changing either time field clears the previous WAV. Audio contexts are closed after decoding and during cleanup, so partial work does not leak between cuts. The Audio Cutter trims in your current browser tab and does not provide fades, normalization, noise reduction, pitch shifting, resampling controls, or remixing. If your cut needs waveform-level fades, lossless compressed-frame cutting, metadata preservation, or production mastering, move the new WAV into a dedicated editor with the original source kept side by side. For a quick, correct range cut that opens cleanly in any DAW or media player, re-running the cut with verified times and a preview pass is the most reliable path.
If you're weighing options, Cut Audio in Your Browser: A First-Time Walkthrough covers this in detail.