A free audio cutter trims a continuous time range from a sound file and hands you a downloadable result without asking for an account, a subscription, or an installer. The output is a freshly encoded PCM16 WAV built from the exact frames your selection covers, not a re-encoded copy of the original compressed stream. Everything happens in the current browser tab: the file is read locally, decoded by the Web Audio API, sliced by sample-frame indices, and exported as interleaved signed 16-bit little-endian PCM with a 44-byte RIFF/WAVE header. Because the cut is performed on decoded samples, the exported duration matches the selected frame count divided by the decoded sample rate, and the file size is that frame count times the channel count times two bytes plus the 44-byte header. Nothing from the source container, including its original codec, bitrate, tags, album art, cue points, and loudness metadata, carries over into the WAV. That scope is the trade-off for a tool that works on every browser-decodable file without a server round-trip and produces a result any audio editor can open.
What "Free Audio Cutter" Really Means
When someone searches for a free audio cutter, they usually want three things at once: no payment, no software to install, and no learning curve. Many "free" tools on the web actually require a signup, place ads on the export, watermark the result, or downgrade the output quality. A genuinely free browser-based cutter should cut the range you specify, return a clean file you can save, and never ask for credentials. The Audio Cutter tool from Lizely fits that description: it works on a single file in the current tab, accepts a start time and an end time in seconds, and produces a downloadable PCM16 WAV. The cost of admission is a browser that supports Web Audio decoding for your file's codec.
The same tool also has to be honest about what it cannot do. A browser-based cutter that only trims a single continuous range is not the same thing as a multi-track editor with fades, normalization, or noise reduction. Anyone trimming a podcast clip, an interview answer, or a single song section can usually get what they need from a range cut. Anyone mastering a song or building a podcast with crossfades, chapter markers, and ID3 tags should expect to export an intermediate file from this tool and finish the job in a dedicated editor.
Why Cut Audio in the Browser Without Uploading
Uploading an audio file to a remote service means trusting that service with the bytes on disk. For a private voice memo, an unreleased demo, or a recording that includes a copyrighted interview, that trust is not always easy to give. A local cutter keeps the file in the browser tab: the file picker reads it, the Web Audio decoder turns it into a buffer of float samples, the cut selects a frame range, and the exporter writes the WAV. The page only uses temporary local URLs to feed the source player and the result download, and it releases those URLs when the file or result is replaced or when the page closes.
The local-only model also explains the explicit safety limits. Because nothing is sent to a conversion server, the tool can enforce a hard ceiling: the encoded source must be no larger than 25 MB, the decoded audio must be no longer than 15 minutes, use no more than eight channels or 192 kHz, and contain no more than 30 million channel-samples in total. An over-limit file is rejected with a clear message instead of being silently shortened or silently transcoded.
Cut an Audio File to a Range Step by Step
- Choose a browser-decodable audio file from your device. The file picker accepts common MP3, WAV, M4A, AAC, Ogg, WebM, and FLAC names and MIME types. The browser still has to actually decode the file, so recognition in the picker does not guarantee playback in your specific browser version.
- Wait for the file to decode. Once Web Audio finishes, the page shows the decoded duration, sample rate, channel count, and frame count. These describe the working buffer the tool will cut from, not necessarily the encoded rate the source file originally used.
- Enter a non-negative start time in moments and a later end time almost instantly. The end value cannot exceed the exact decoded duration and must be strictly greater than the start. Both values are converted to the nearest decoded sample-frame boundary; a selection that rounds to less than one frame is rejected instead of producing an empty file.
- Select Cut and export WAV. The result replaces any previous cut, the prior download URL is released, and the page displays the output duration and file size so you can confirm before downloading.
- Preview the resulting PCM16 WAV to check the spoken words or musical transient at both boundaries, then save the file to your device. Changing either time field clears the previous WAV, and cutting again replaces and releases the prior download URL.
How the Cut Maps to Decoded Sample Frames
The cut is not driven by a wall-clock timer or a re-encoded packet index. Start and end times in moments are converted to frame indices using the AudioBuffer sample rate, and the cut pulls every decoded sample between those indices. The start frame is included and the end frame is excluded, mirroring how slicing works on a frame array. This is why the production exporter and any isolated test of the cut share the same function: the displayed cut duration is the selected frame count divided by the decoded sample rate, not an estimate from a separate clock.
Two consequences follow. First, Web Audio may resample the source while decoding so that it matches the AudioContext sample rate. The sample rate you see in the tool's readout therefore describes the decoded working buffer and the WAV that comes out of it, not necessarily the encoded rate the file shipped with. Second, because the cut is performed on float samples, the exporter has to convert them to integer PCM. Any sample value below negative full scale is clamped to -32768, any value above positive full scale is clamped to 32767, and any non-finite value (NaN, plus or minus Infinity) is written as silence. Multiple channels stay in their original order and are interleaved in the output.
File size for the exported WAV follows from the same numbers. Each selected frame contributes two bytes per channel, and a 44-byte RIFF/WAVE header sits at the front. A five-second stereo cut from a 44.1 kHz decoded buffer is 5 x 44,100 = 220,500 frames; 220,500 x 2 channels x 2 bytes = 882,000 bytes; plus 44 bytes for the header gives 882,044 bytes, or roughly 861 KB. The tool reports the exact output size before download, so you do not have to do the arithmetic yourself.
What the PCM16 WAV Output Contains and What It Drops
The WAV you download is a new file, not a re-tagged copy of the original. The encoded MP3, AAC, Vorbis, Opus, or FLAC payload is gone, replaced by uncompressed PCM samples. The file does not retain the source's tags, album art, cue points, chapters, loudness metadata, encoder settings, or other container fields. That trade-off is the price of a single uniform output format that every audio editor can open.
| Preserved in the PCM16 WAV | Dropped from the WAV |
|---|---|
| Decoded audio samples in original channel order | Original codec (MP3, AAC, Vorbis, Opus, FLAC, and so on) |
| Decoded sample rate used during Web Audio decode | Source bitrate and original encoder settings |
| Channel count, up to the eight-channel ceiling | ID3, Vorbis comments, album art, embedded artwork |
| 16-bit signed little-endian PCM sample values | Cue points, chapters, and loudness metadata |
| 44-byte RIFF/WAVE header with exact data size | Any other container fields the source carried |
If your downstream tool needs a compressed file, an MP3 or AAC encoder is a separate step. You can still use the WAV as the source for that encoder, because the decoded samples in the WAV match the cut you selected.
File Size, Limits, and Supported Input Formats
Because PCM is uncompressed, a short MP3 source can expand into a much larger WAV. The output data size is the selected frame count multiplied by the channel count and two bytes per sample, plus a 44-byte header. A ten-minute stereo segment at a decoded 48 kHz sample rate is 600 x 48,000 x 2 x 2 = 115,200,000 bytes, or about 110 MB. That sits well inside the tool's processing limits, but it is worth noting if you are short on disk space or planning to email the result.
| Limit | Maximum value |
|---|---|
| Encoded source file size | 25 MB (25 MiB) |
| Decoded duration | 15 minutes |
| Channel count | 8 channels |
| Decoded sample rate | 192 kHz |
| Decoded channel-samples across all channels | 30,000,000 |
The decoded-sample limit matters because high-rate and multichannel files reach it before they reach the duration limit. A 192 kHz stereo file hits the 30-million cap at about 78 seconds of decoded audio. The browser still has to decode the full source to inspect the buffer, so a highly compressed long file can still use temporary memory during the decode attempt even if the decoded result will eventually be rejected.
Input support depends on the codecs your browser ships with. The picker recognizes MP3, WAV, M4A, AAC, Ogg, WebM, and FLAC names and MIME types, but Safari, Chrome, Firefox, and different operating systems ship different decoders. An unsupported, damaged, mislabeled, encrypted, or incomplete file produces a clear decode error instead of a silent substitution, and the tool does not install codecs or send the file to a conversion server to work around the gap. The MDN reference for decodeAudioData lists the codecs each browser supports.
When a Dedicated Editor Is the Better Choice
A frame-accurate cut is enough for a lot of jobs: trimming dead air from a podcast, clipping a song for a ringtone, lifting one answer from an interview, or pulling a single effect from a sound library. The free audio cutter handles those cases with no upload and no install. It also gives you a clean, uncompressed WAV you can re-encode or re-edit later.
For more involved jobs, plan to move to a desktop editor. Waveform-level fades, gain ramps, noise reduction, EQ, and compression all sit outside this tool's feature set. The tool does not offer a target resampling control, a remix control, normalization, fade-in or fade-out, pitch shifting, or output compression. It also cannot perform compressed-frame-aware cuts that respect MP3 or AAC packet boundaries the way a decoder-aware splitter does; the cut is always on the decoded sample buffer, and the WAV does not retain the source codec. Metadata preservation, including tags and album art, is intentionally not part of the output. If any of those matter for the final deliverable, treat the WAV from this tool as an intermediate and finish the work in a dedicated editor.