A failed cut almost always traces back to one of three places: an unsupported codec the browser cannot decode, a time range that snaps to zero frames after rounding, or a safety limit on the decoded audio buffer. The Audio Cutter reads and decodes every file inside your current browser tab using the Web Audio API, so nothing uploads, and every error points to the specific limit that stopped the cut. Because the decode happens before any timing is captured, the page can display the exact decoded duration, sample rate, channel count, and frame count you can legally cut against. The export always writes a fresh RIFF/WAVE file with interleaved 16-bit PCM samples and a 44-byte header, so the download will never carry over your source's MP3, AAC, Vorbis, Opus, or FLAC encoding, tags, or metadata. Troubleshooting therefore comes down to reading the page's decoded stats, picking a valid frame range inside them, and confirming the preview before you save the file.

What the Audio Cutter Does Before the Cut
When something goes wrong, it helps to know what the tool is doing at each stage. The file picker recognizes common MP3, WAV, M4A, AAC, Ogg, WebM, and FLAC names and MIME types, but recognition does not guarantee that every variation of those codecs can actually be decoded by your current browser. Container and codec support differs across Safari, Chrome, Firefox, operating systems, and browser versions, so the same filename can decode in Chrome and fail in Safari. After you choose a file, the browser validates the size against the 25 MB encoded limit and passes the bytes to Web Audio's decodeAudioData function. Web Audio may resample the source to the AudioContext sample rate while decoding, which means the displayed sample rate describes the decoded working buffer and the exported WAV, not necessarily the encoded rate inside your original file. The decoded AudioBuffer then drives everything else: the displayed duration, the channel count, the frame count, the time entry rounding, and the exported WAV. If decoding fails, the page surfaces a clear decode error and stops, because no valid buffer exists to cut against. Audio contexts are closed after decoding and during cleanup, so no leftover playback state lingers between files.
Common Problems and What They Mean
Most reader complaints fall into a small number of buckets. The table below pairs the symptom you see with the underlying reason inside the Audio Cutter, so you can decide which section to jump to.
| Symptom | Likely Cause | Where to Look |
|---|---|---|
| Page shows a decode error and never displays stats | Browser cannot decode the codec, or the file is damaged, mislabeled, encrypted, or incomplete | Try a different browser or re-export the file from a trusted source |
| Stats appear but Cut rejects the time range | End less than or equal to start, start negative or at the end, end past the decoded duration, or selection rounds to zero frames | Re-enter the times and watch the displayed cut duration |
| Downloaded file is much larger than the source | Output is uncompressed PCM16 WAV, not the original compressed codec | Expected for short clips; compress later if needed |
| Sample rate on the page does not match the source's tag | Web Audio resampled to the AudioContext rate during decoding | Treat the displayed rate as the export rate |
| File is rejected by a safety limit | File over 25 MB before decoding, or decoded audio over 15 minutes, 8 channels, 192 kHz, or 30 million channel-samples | Trim or downmix the source in another tool first |
Fix File and Codec Decode Errors
A decode error means the browser reached your file, started passing bytes to Web Audio, and the decoder rejected them. Common triggers are DRM-protected downloads, recordings saved with an unusual codec variant, files truncated during transfer, and M4A or MP4 containers using a codec the current browser cannot read. The page does not install codecs, does not send the file to a conversion server, and does not silently substitute another source, so the only fixes are at the file level. Re-export the audio from the original application using a widely supported codec such as MP3 at 128 kbps or higher, or WAV PCM. Verify the new file plays in a desktop player before returning to the Audio Cutter. If the file plays locally but still fails in the tool, the codec container is probably the issue: try changing the file extension from .m4a to .mp4 or vice versa, or open and re-save the file in Audacity, ffmpeg, or another trusted editor. Switching browsers is the fastest test, since Safari, Chrome, and Firefox maintain different codec tables.
Fix Time Entry and Frame Range Errors
Start and end times are converted to the nearest decoded sample-frame boundary using the AudioBuffer sample rate, so the page's displayed cut duration is always the selected frame count divided by the decoded sample rate, never a free-running timer. The start frame is included and the end frame is excluded. End must be greater than start, start cannot be negative or at the end of the audio, and end cannot exceed the exact decoded duration. A selection that rounds to less than one frame after that conversion is rejected rather than producing an empty file. If your decimal input produces zero frames because the rounded difference is smaller than one sample period, add a tenth of a second to either end. If you are cutting from the very end of a long file, use the page's displayed duration as the upper bound instead of a value from a tag editor, since the displayed duration is the exact post-decode value. Changing either time field clears the previous WAV, and cutting again replaces and releases the prior download URL, so always preview after a new entry.
Safety Limits That Reject the Cut
The Audio Cutter applies explicit safety limits rather than silently shortening or resampling to fit. The encoded input file must be no larger than 25 MB. After decoding, audio must be no longer than 15 minutes, use no more than eight channels or 192 kHz, and contain no more than 30 million samples across all channels. The decoded-sample limit means a high-rate or multichannel file may reach the processing boundary before the duration limit. Nothing is shortened automatically: an over-limit file is rejected with a message. Because decoding has to happen before decoded duration and sample count can be inspected, a highly compressed long file may still consume temporary memory during the decode attempt. The cleanest workaround is to trim or downmix the source in a desktop editor first, then return to the tool with a smaller file.
Verify the Result After Cutting
The Audio Cutter reports the output duration and file size before download, which lets you sanity-check the result without playing it. You can also estimate the data size with the rule the page uses: output data size in bytes equals the selected frame count multiplied by the channel count and two bytes per sample, plus a 44-byte RIFF header. For example, a 10-second stereo selection at a 44.1 kHz decoded rate works out to 10 × 44,100 = 441,000 frames, then 441,000 × 2 channels = 882,000 channel-frames, then 882,000 × 2 bytes = 1,764,000 bytes of PCM data, plus 44 bytes for the header, for a total file size of 1,764,044 bytes. If the page reports a different value, redo the cut. Confirm the spoken word or musical transient at both boundaries by listening to the preview, because the export uses the same frame-range function as the isolated tests the tool runs. For dependable work, keep the original file and reach for a dedicated editor whenever you need waveform-level fades, lossless compressed-frame cutting, metadata preservation, or production mastering. If the output still does not sound right after these checks, see how to fix a result that looks wrong after you cut audio for a deeper walk-through.
Troubleshoot and Cut Audio in Your Browser
- Open the Audio Cutter and choose a browser-decodable file that is at most 25 MB. Wait for Web Audio to finish decoding it inside your current tab.
- Read the page's decoded duration, sample rate, channel count, and frame count. If those numbers never appear, the decode failed: re-export the file from a trusted editor or try a different browser.
- Enter a non-negative start time in seconds that is strictly less than the decoded duration. Enter an end time that is greater than start and does not exceed the decoded duration.
- Watch the displayed cut duration update as you type. If the cut duration drops to zero, the rounded selection is smaller than one frame, so widen the range by at least a tenth of a second on either side.
- Select Cut and export WAV, then preview the PCM16 result. Confirm the spoken word or musical transient at both the start and end boundaries.
- Compare the page's reported output size against the frame × channels × 2 + 44 formula. If the values match, click download to save the new RIFF/WAVE file.
- If the cut is rejected, check the file against the safety limits (25 MB encoded, 15 minutes decoded, eight channels, 192 kHz, 30 million channel-samples), trim or downmix the source, and try again with a smaller input.
For a deeper look, see Details That Matter When You Cut Audio.