Audio Trimmer & MP3 Converter
Trim audio and convert to MP3 without installing anything. Drag the waveform to cut a precise clip, export MP3 or WAV, normalise the level, add fades and change speed — entirely on your device.
How to use Audio Trimmer & MP3 Converter
- Load an audio file — MP3, WAV, M4A, OGG and more.
- Drag the waveform handles to select the section you want.
- Adjust gain, fades, speed and normalisation.
- Export as MP3 or WAV and download.
About Audio Trimmer & MP3 Converter
Cutting thirty seconds out of a recording should not mean installing a digital audio workstation, creating an account, or watching an upload bar. Audio Studio covers the operations people genuinely reach for — trim to a section, change the format, fix the level, fade the ends, slow it down — and it does them against a drawn waveform, so you drag the handles to where the speech actually starts instead of guessing at timecodes and listening again to check.
Underneath, the file is decoded to raw samples by the browser and every edit is applied by rendering that audio through an offline graph: the trimmed region plays into a gain node whose value is automated for the fades, with the channel count and sample rate fixed at the output. Nothing is processed in real time, so a ten-minute file renders far faster than it plays. MP3 writing is handled by an encoder compiled to run inside the page, which is what keeps voice notes, interview recordings and unreleased mixes off anybody else’s servers.
Peak normalisation is the setting most worth understanding. It finds the loudest sample in the finished clip and scales the whole file so that peak just reaches the ceiling — every quiet passage rises by exactly the same amount, leaving the dynamics untouched. That fixes the common problem, which is a recording made with the input gain set too low. It is not the same as compression or loudness matching: if a clip is quiet only because one cough is loud, normalising will barely move it, and the real fix is to trim the cough out first.
Three caveats are worth stating. Speed is changed by resampling, so pitch moves with it — 1.25× makes a voice higher as well as faster, which is fine for skimming a lecture and wrong for music. Which input formats work depends on your browser’s own decoder rather than on this page: MP3 and WAV are universal, while M4A, AAC and FLAC are widely but not perfectly supported. And MP3 is lossy, so re-encoding one always costs a little; export WAV when the file is heading into further editing, and keep MP3 for the finished version, where 192 kbps is transparent enough for speech and 256–320 kbps is the sensible range for music.
Frequently asked questions
Is my audio uploaded to a server?
Never. Decoding, editing and encoding all happen locally using the Web Audio API and an in-browser MP3 encoder. A confidential interview or an unreleased track can be cut here without a copy existing anywhere but your own machine.
What formats can I import and export?
Export is MP3 at 96 to 320 kbps, or WAV at 16-bit or 24-bit. On the way in, anything your browser can decode is accepted — MP3 and WAV everywhere, with M4A, AAC, OGG, FLAC and Opus depending on the browser you are using.
Does changing the speed also change the pitch?
Yes. Speed is applied by resampling, so a clip at 1.25× is both faster and a little higher. That suits skimming a lecture or a podcast; it is the wrong approach for music, which needs pitch-preserving time-stretching.
What does normalise actually do?
It scales the entire clip so its loudest sample just reaches the ceiling, lifting quiet passages by exactly the same amount as loud ones. It rescues a recording made with the gain too low, but it will not even out a track whose levels swing about — that needs compression.
Which MP3 bitrate should I choose?
For speech, 128 or 192 kbps is indistinguishable from the source and keeps files small. For music, 256 or 320 kbps is the sensible range. Going higher than the original was encoded at recovers nothing, since the detail was discarded before you opened the file.
Can I trim audio on a phone?
Yes — the waveform handles are draggable by touch and playback works as usual. Long recordings are the limiting factor rather than the interface, because the whole file is decoded to raw samples in memory before editing begins.
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