Audio File Info
Processed locally in your browser — nothing is uploaded.
Quick answer
This tool reads an audio file’s core specs — duration, bitrate, sample rate, channels and bit depth — and shows them instantly. Unlike the other audio tools it uses the browser’s built-in Web Audio API rather than the ffmpeg engine, so there is **no 10 MB engine download and no waiting**: the moment you drop a file, its numbers appear. Everything is read on your device; nothing is uploaded.
How to audio file info
Drop an audio file
Drag any common audio file — MP3, WAV, M4A, FLAC, OGG — onto the drop zone. No engine loads first.
Read the specs instantly
The browser decodes the header locally and shows duration, sample rate, channel count and the estimated bitrate right away.
Use the numbers
Use the specs to pick a conversion target, diagnose a too-large file, or confirm a recording is the sample rate you expected.
Why is this tool instant when the others download an engine?
The rest of the audio suite needs to *transform* audio — cut it, convert it, re-encode it — which requires ffmpeg compiled to WebAssembly, a roughly 10 MB payload downloaded once and cached. Reading a file’s properties needs none of that. Every modern browser ships the Web Audio API, which can decode an audio file and report its sample rate, channel count and duration natively. This tool uses exactly that, so there is nothing to download and nothing to wait for — the specs appear the instant you drop the file.
What do the numbers actually mean?
The specs describe how the sound was captured and stored. Here is what each one tells you:
| Property | What it means | Typical values |
|---|---|---|
| Duration | Length of the audio in time | e.g. 3:47 |
| Sample rate | Audio snapshots captured per second | 44,100 Hz (CD) · 48,000 Hz (video) |
| Channels | Separate audio streams | 1 = mono · 2 = stereo |
| Bit depth | Resolution of each sample (PCM/WAV) | 16-bit (CD) · 24-bit (studio) |
| Bitrate | Data used per second of audio | 128–320 kbps (MP3) · ~1,411 kbps (CD WAV) |
What is sample rate, and why 44.1 vs 48 kHz?
Digital audio works by taking rapid snapshots — samples — of a sound wave. Sample rate is how many of those snapshots are taken per second, measured in hertz. A higher rate captures higher frequencies faithfully. Two standards dominate: 44.1 kHz is the CD and music standard, while 48 kHz is the standard for video and film. That is why audio pulled from a video is usually 48 kHz and audio from a music library is usually 44.1 kHz — and mixing the two without resampling can cause sync drift in a video project.
What are bit depth and channels?
Bit depth is how finely each sample is measured — 16-bit gives CD-quality dynamic range, while 24-bit is used in studios for more headroom while recording. It applies to uncompressed PCM formats like WAV; lossy formats like MP3 describe their quality with bitrate instead. Channels is simply how many separate streams the file carries: one for mono, two for stereo (left and right). If a file reports one channel and you expected two, it was recorded or exported as mono — the mono to stereo tool can duplicate it to both sides.
How is the bitrate worked out?
Bitrate is the amount of data spent on each second of audio, and it is the single biggest lever on an MP3’s file size and quality — 320 kbps sounds better and weighs more than 128 kbps. For a constant-bitrate file it is read directly; otherwise it is estimated from the file size and duration, which is why a variable-bitrate MP3 may show an average. If a file is bigger than you want, that number tells you how much room there is to compress audio.
Is my file uploaded to read its info?
No. The browser decodes the file’s header locally through the Web Audio API and reads the numbers off it — the audio never leaves your device and never reaches a server. Open DevTools → Network and drop a file: no upload request is made. Since there is not even an engine to fetch, this tool makes essentially no network activity at all.
Related audio and media tools
Once you know the specs, act on them: convert the audio to another format, compress it if the bitrate is higher than you need, or edit its tags. If the file is the soundtrack of a captioned video, you can turn its subtitle file into plain text for a transcript.
Your files never leave your device
Processed locally in your browser — nothing is uploaded. You can verify it yourself: open your browser’s developer tools, watch the Network tab, and use the tool — no upload request is ever made.
Frequently asked questions
Why does this tool load instantly when others make me wait?
It reads file properties with the browser’s built-in Web Audio API instead of the ffmpeg engine, so there is no ~10 MB download. The specs appear the moment you drop a file.
What sample rate should my audio be?
Use 44.1 kHz for music and 48 kHz for anything going into video or film. Matching the project’s rate avoids resampling and sync drift.
What is the difference between mono and stereo here?
Channels tells you: 1 channel is mono (a single stream) and 2 is stereo (separate left and right). If you expected stereo but see 1, the file is mono.
Why does my MP3 not show a bit depth?
Bit depth is a property of uncompressed PCM formats like WAV. Lossy formats such as MP3 and AAC express their quality as a bitrate instead, so bit depth is not reported for them.
Is the bitrate exact?
For constant-bitrate files it is read directly. For variable-bitrate files it is estimated from size and duration, so it represents the average bitrate.
Which formats can it read?
Anything the browser can decode — MP3, WAV, M4A/AAC, FLAC, OGG and more. The Web Audio API handles the common formats natively.
Is my file uploaded to read its info?
No. The header is decoded locally in your browser and nothing is sent to a server. With no engine to fetch either, the tool makes essentially no network requests — confirm it in the Network tab.