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Audio File Info
Drop in one or many audio files to see what is really inside: the codec, bitrate and whether it is constant or variable, sample rate, bit depth, channels, exact length, every tag and the embedded cover art. It also measures peak and RMS levels, and exports the whole table as CSV. Files are read on your device and never uploaded.
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No files yet. Add one or more audio files to see their technical details side by side, or press Try a sample file to see a demo report.
| File | Format | Codec | Bitrate | Mode | Sample rate | Bit depth | Channels | Duration | Size | Peak | RMS | Title / Artist |
|---|
No cover art
File
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How to use the Audio File Info
- Add audio files: drop them on the box or click to choose. You can add many at once, or press Try a sample file to see a demo.
- Read the table: each file gets a row with format, codec, bitrate, mode, sample rate, bit depth, channels, length, size, peak and RMS.
- Select a row to open the full report: stream details, encoder information, every tag, cover art, levels and a waveform overview.
- Check the notes: warnings appear for truncated files, missing VBR headers, ringtones over 40 seconds and similar problems.
- Export: press Export CSV for the whole table, Copy table to paste into a spreadsheet, or Download JSON for one file.
What each value means
| Field | What it tells you | Typical values |
|---|---|---|
| Container / format | The file wrapper. The same codec can sit in different containers (AAC in .m4a, .m4r or .aac; Opus in .ogg, .opus or .webm). | MP3, WAV, FLAC, M4A, Ogg, WebM |
| Codec | How the sound is compressed, or PCM when it is not compressed. | MPEG Layer III, AAC-LC, Opus, Vorbis, FLAC, ALAC, PCM |
| Bitrate | Average audio data per second. For PCM it is sample rate × bit depth × channels. | 128 to 320 kbps (MP3), 96 to 256 kbps (AAC), 1,411 kbps (CD-quality WAV) |
| Mode | CBR keeps the bitrate constant, VBR varies it with the music, ABR aims for an average. | CBR, VBR, ABR |
| Sample rate | Samples per second per channel. It sets the highest frequency that can be stored (half the rate). | 44.1 kHz (music), 48 kHz (video), 96 kHz (hi-res) |
| Bit depth | Bits per sample in PCM and lossless files. Each bit adds about 6 dB of dynamic range. | 16-bit (CD), 24-bit (studio), 32-bit float |
| Peak | The loudest single sample, in dB relative to full scale. | −1 to 0 dBFS for mastered music |
| RMS | The average power of the whole file. Higher RMS sounds louder. | −20 to −8 dBFS for music |
The formulas used
- PCM bitrate = sample rate × bit depth × channels. Example: 44,100 × 16 × 2 = 1,411,200 bit/s = 1,411.2 kbps.
- Duration of an MP3 = number of audio frames × samples per frame ÷ sample rate. MPEG-1 Layer III frames hold 1,152 samples; MPEG-2 and 2.5 Layer III frames hold 576.
- Average bitrate of a compressed file = audio bytes × 8 ÷ duration. For Ogg and WebM the whole file size is used, so container overhead is included.
- Peak (dBFS) = 20 × log10(largest |sample|). RMS (dBFS) = 20 × log10(√(Σ sample² ÷ N)). On this unweighted scale a full-scale sine wave measures −3.01 dBFS RMS.
- Crest factor = peak dB − RMS dB. Heavily compressed modern masters often show 8 dB or less; dynamic recordings 15 dB or more.
Tips for reading the results
- Upscaled files: a high bitrate or 24-bit depth does not guarantee quality. A file converted from a lower-quality source keeps the original's limits.
- Clipped samples at full scale usually mean the audio was pushed too hot during mastering or conversion and may sound distorted.
- Dual mono means the left and right channels are identical; saving it as mono halves the size with no loss.
- VBR MP3s without a Xing or VBRI header can show a wrong length in some players. Re-encoding or re-tagging with a modern tool adds the header.
- iPhone ringtones (.m4r) must be 40 seconds or shorter; the tool flags longer files.
- Loudness for streaming is measured in LUFS, which weights frequencies the way people hear them. Use a LUFS meter when you need a platform loudness target; RMS here is a quick, unweighted level.
Frequently asked questions
How can I tell if an MP3 is really 320 kbps?
The tool reads every MPEG frame, so it shows the true bitrate and whether it is constant (CBR) or variable (VBR). A 320 kbps label does not prove quality, though: a file re-encoded from a 128 kbps source keeps that source's missing treble. To check, open it in a spectrum analyzer: real high-bitrate MP3s usually reach 19 to 20 kHz, while upscaled ones often cut off near 16 kHz.
What is the difference between CBR and VBR?
CBR (constant bitrate) uses the same number of bits for every frame, giving a predictable file size. VBR (variable bitrate) spends more bits on complex passages and fewer on simple ones, usually giving better quality for the same size. ABR (average bitrate) is VBR that aims for a target average.
Why is bit depth shown as n/a for MP3, AAC and Opus?
Lossy codecs do not store samples at a fixed bit depth; they store compressed frequency data that the decoder turns into floating-point samples. Bit depth only applies to PCM and lossless formats such as WAV, AIFF, FLAC and ALAC.
How are peak and RMS levels measured?
The file is decoded by your browser at its native sample rate. Peak is the largest absolute sample value, and RMS is the square root of the mean of all squared samples, both in dB relative to digital full scale (dBFS). A full-scale sine wave reads 0 dBFS peak and about −3.01 dBFS RMS on this scale. A decoded lossy file can peak slightly above 0 dBFS.
Why does Opus always show a 48 kHz sample rate?
Opus always encodes and decodes at 48 kHz internally. The original input sample rate stored in the header is shown separately, but players output Opus at 48 kHz.
Are my audio files uploaded anywhere?
No. Files are read with JavaScript on your device and decoded with your browser's built-in Web Audio API. Nothing is sent to a server, so it also works for private recordings.
Why does my WebM recording show no duration in the header?
Recordings made by browsers with MediaRecorder are written as a live stream, so the length is often missing from the header. The tool then takes the length from decoding the audio.
Are all the tools on FreeToolCabin tested for quality and accuracy?
Yes! Every tool on our site undergoes automated Q/A testing in real browser environments to ensure high precision, correct output, and zero errors. If you happen to spot a bug or have a suggestion, please contact us at [email protected] and our team will review it right away.
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