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Lizely

Audio Cutter

Cut an exact time range from browser-decodable audio and download a local PCM16 WAV.

Privacy: your files never leave your device. All processing happens locally in your browser.

How to use

  1. 1.Choose a browser-decodable audio file within the displayed file and decoded-audio limits.
  2. 2.Enter a non-negative start time and a later end time that does not exceed the decoded duration.
  3. 3.Select Cut and export WAV, preview the PCM16 result, then download it.

About Audio Cutter

Audio Cutter trims one continuous range from an audio file entirely in the current browser. Choose a file, wait for Web Audio to decode it, enter start and end times in seconds, and create a downloadable result. The page displays the decoded duration, sample rate, channel count, and frame count before cutting. The original file and decoded samples are not uploaded to Lizely. Temporary local URLs support the source player and result download, and they are released when a file or result is replaced or when the page closes.

Input support depends on the 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. The tool does not install a codec, send the file to a conversion server, or silently substitute another source.

Start and end times are converted to the nearest decoded sample-frame boundary using the AudioBuffer sample rate. Web Audio may resample the source to the AudioContext sample rate while decoding, so the displayed sample rate describes the decoded working buffer and the exported WAV, not necessarily the source file's encoded rate. 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 is rejected rather than producing an empty file. The production exporter and isolated tests share the same frame-range function, so displayed cut duration is the selected frame count divided by the decoded sample rate instead of an unrelated timer estimate.

Every download is a newly encoded RIFF/WAVE file containing interleaved, little-endian, signed 16-bit PCM samples. It does not retain the source's MP3, AAC, Vorbis, Opus, FLAC, or other compressed encoding. It also does not preserve tags, album art, cue points, chapters, loudness metadata, encoder settings, or other container fields. Floating-point Web Audio samples are limited to the PCM range from -1 through 1; negative full scale becomes -32768, positive full scale becomes 32767, and non-finite values are written as silence. Multiple channels remain separate and are interleaved in their original channel order.

Because PCM WAV is uncompressed, the result can be much larger than a short MP3 or AAC source. Its data size is the selected frame count multiplied by the channel count and two bytes per sample, plus a 44-byte WAV header. The tool reports the output duration and file size before download. It does not offer a target resampling control, remix, normalize, fade, remove noise, change pitch, or compress the WAV. The export uses the decoded AudioBuffer rate after any resampling performed by Web Audio. If a decoded source is stereo, the result stays stereo; if it has more channels, those channels are retained up to the stated eight-channel limit. Playback software must support the resulting channel layout.

Safety limits are explicit rather than silent. 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 high-rate or multichannel files may reach the processing boundary before the duration limit. Nothing is shortened automatically: an over-limit file is rejected with a message. 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.

Changing either time field clears the previous WAV, and cutting again replaces and releases the prior download URL. Choosing another file clears the old decoded buffer, players, errors, and result. In-flight reads and decodes are guarded so an older job cannot overwrite a newer selection. Audio contexts are closed after decoding and during cleanup. For dependable work, preview the cut WAV, confirm the spoken words or musical transient at both boundaries, keep the original file, and use a dedicated editor when you need waveform-level fades, lossless compressed-frame cutting, metadata preservation, or production mastering.

Methodology & sources

The browser validates the selected file name or MIME type and a 25 MiB byte limit, then Web Audio decodes it and may resample it to the AudioContext rate. Decoded audio is rejected above 15 minutes, eight channels, 192 kHz, or 30 million channel-samples. Start and end seconds are rounded to nearest frame indices at the decoded AudioBuffer rate; the start is inclusive and end exclusive. Channel Float32 samples in that range are clamped to -1 through 1 and encoded in original channel order as interleaved signed little-endian PCM16. A 44-byte RIFF/WAVE header records PCM format 1, channel count, decoded sample rate, byte rate, block alignment, 16 bits per sample, and exact data size.

Frequently asked questions

Does the download keep the original MP3 or AAC encoding?
No. Every result is a newly encoded, uncompressed PCM16 WAV. The original compressed codec, bitrate, and metadata are not preserved.
Why can the cut WAV be larger than the source?
PCM16 WAV stores uncompressed samples. Output data uses two bytes for every selected frame in every channel, plus a 44-byte header.
Which input formats work?
The picker recognizes common MP3, WAV, M4A, AAC, Ogg, WebM, and FLAC files, but actual decoding depends on the codecs supported by your current browser.
Are audio files uploaded?
No. File reading, Web Audio decoding, frame selection, WAV encoding, preview, and download all happen in the current browser tab.

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