Audio Converter
How to Use
📖 Tool Introduction
A converter that changes audio formats right in the browser, with no ffmpeg to install and no files sent to a server. After uploading you get a waveform, you can drag the handles at either end to pick just a segment to convert, and you can queue several files for batch processing.
Some twenty output formats are on offer, including the common MP3, WAV, OGG, M4A, AAC, OPUS and FLAC, plus special-purpose ones like AIFF, AU, CAF and the M4R ringtone format. Sample rates run from eight kilohertz up to ninety-six, bit depth, bitrate and channel count are all settable independently, and there are volume normalisation, fade in and out, speed change and reverse effects.
The audio encoding engine is fetched from an online resource, so the first conversion needs a connection, and a failed load gives a clear message with a retry button. The audio itself never leaves your device: decoding and encoding both happen locally.
Most people reach for this for a handful of jobs: trimming a podcast episode down and compressing it to MP3 for distribution, shrinking a WAV from a voice recorder into something smaller, pulling the soundtrack out of a video as a standalone audio track, or cutting a short ringtone for a phone. Each of those used to mean installing something or opening a terminal; here you drop the file in, pick your settings and get a file back.
One point about privacy is worth stating plainly. Decoding and encoding both happen on your own machine, and the audio is never sent to a server, so work material or meeting recordings that should not leave your hands are safe to process here. The trade-off is that speed depends on your hardware, so allow a little time for long recordings.
Before converting, decide what the file is for. Keep FLAC or WAV when archiving something you care about, choose MP3 or AAC when sending it to someone to listen to, and consider OGG or OPUS for web assets. Leaving the sample rate as it is usually the safest choice unless you specifically want telephone-grade quality or a high resolution master, and changing channels from stereo to mono roughly halves the size for speech recordings where stereo carries no real information.
✨ Key Features
- Over twenty output formats including MP3, WAV, OGG, FLAC, M4A, AAC and OPUS alongside AIFF, AU, CAF and M4R
- Sample rates from eight to ninety-six kilohertz, with sixteen, twenty-four and thirty-two bit float depths
- Mono and stereo conversion, with bitrates from sixty-four to three hundred and twenty kilobits per second
- Draggable handles on the waveform to select a range, backed by start and end time fields for precise cutting
- Volume normalisation, fade in and out, playback speed adjustment and audio reversal
- A batch mode that queues several files and runs them all under one parameter set
🚀 How to Use
- Click the upload area to pick an audio file, or drag one in; over twenty input formats are accepted
- Choose the target format, then set sample rate, bit depth, bitrate and channels
- For trimming, use the effects tab for volume and fades, and the crop tab to drag a range on the waveform
- With several files, switch to the batch tab and add them under one shared setting
- Press convert and download the finished files when it completes
⚠️ Things to Note
- The encoding engine loads over the network: The first conversion fetches encoding resources online. A failed load shows a message with a retry, and offline you cannot convert
- Lossy conversion is one-way: Moving from a lossless format to MP3, OGG or similar discards information that cannot be recovered afterwards
- Large files take a while: Conversion runs in the browser, so long recordings or high sample rates can use a lot of memory and time. Give it a moment
- Browsers vary in format support: Encoding for a few formats leans on native browser capability, so individual browsers may not support every output listed
- Ringtones have length limits: When exporting M4R, check the duration your target device allows, since over-long files may be rejected as ringtones