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Audio Quality Check

Drop a track to measure levels, spot clipping and quiet parts, and see what to fix next — all on your device.

Audio tool

Analyze audio quality

Runs in your browser

Drop an audio file here

or click to choose from your computer

MP3 · WAV · FLAC · AAC · M4A · OGG

✓ Local processing in your browser · Files are not uploaded to our servers

About Audio Quality Check

Audio Quality Check decodes a file in your browser and gives it a technical health check: sample peak, true peak, RMS, integrated loudness in LUFS, loudness range, clipping, DC offset, stereo phase, silence at the edges, and a level graph you can hover over. It then turns those numbers into plain recommendations and shows how many dB you would have to add or remove to hit common streaming targets.

It is built for the moment before you publish or process a file: a podcast episode that sounds quiet next to other shows, a mix that distorts after MP3 export, a voice memo with a long dead intro, a song that feels different on the phone speaker. Instead of guessing, you see which number is off and which tool fixes it. Nothing is uploaded — analysis runs on your device, so unreleased tracks and private recordings stay private.

One honest limit: the tool measures levels and signal health, not codec quality. It cannot tell you the bitrate of an MP3 or whether a file was upscaled from a lower-quality source.

How to check a file

01

Add audio

Drop an MP3, WAV, FLAC, M4A, OGG or similar file, or open one from Your Library. Dropping a new file on the report re-runs the analysis.

02

Read the report

Check the four headline numbers first — peak, RMS, loudness and true peak — then hover the level graph to find the exact moments that are too loud, too quiet or silent.

03

Fix and re-check

Follow the recommendations to Volume Booster, Trimmer or Equalizer, download the result, then drop it back in here to confirm the numbers moved the way you wanted.

What the numbers mean

  • Sample peak (dBFS) — the highest stored sample, where 0 dBFS is digital full scale. Staying below about −1 dBFS leaves room for encoding.
  • True peak (dBTP) — the highest level of the reconstructed waveform between samples, estimated with 4× oversampling. It can be higher than the sample peak, and it is the number lossy encoders and DACs actually react to. Keep it at or below −1 dBTP.
  • RMS (dBFS) — average signal energy with no weighting. Handy for comparing the body of two files, but bass-heavy material reads higher than it sounds.
  • Loudness (LUFS) — integrated loudness using K-weighting (which mimics how the ear hears frequencies) and gating (which ignores near-silent passages). This is the figure streaming platforms normalize to.
  • Loudness range (LU) — the spread between the quiet and loud parts, measured over 3-second windows. Speech is usually low, orchestral music high; very large values in a podcast mean listeners will keep reaching for the volume knob.
  • Crest factor (dB) — peak minus RMS. A small value means a dense, compressed signal; a large one means sharp transients over a soft body.
  • Stereo correlation — +1 means both channels are identical, 0 is wide and uncorrelated, and negative values mean phase cancellation when the file is played in mono.
  • DC offset — a constant shift of the waveform away from zero. It wastes headroom and can cause clicks where audio is cut.

Loudness targets cheat sheet

Platforms turn loud files down to a reference level so every track plays at a similar volume. These are the commonly cited values — services change their policies, so treat them as a guide and check the current specification when it matters.

DestinationLoudness targetTrue-peak ceiling
Spotify, YouTube, Tidalabout −14 LUFS−1 dBTP
Apple Musicabout −16 LUFS−1 dBTP
Podcasts−16 LUFS (stereo)−1 dBTP
EBU R128 broadcast−23 LUFS−1 dBTP
ATSC A/85 broadcast−24 LKFS−2 dBTP

The report's “Gain to reach platform targets” card does this arithmetic for your file: it subtracts your measured loudness from each target and warns you when the extra gain would push true peak past the ceiling.

Common problems and what to do

Peaks near 0 dB, flat tops

Runs of full-scale samples mean the waveform was already clipped. Lower the level before any lossy export, and if possible go back to an earlier, unclipped version — clipping cannot be undone.

Quiet voice recording

Speech often measures −30 LUFS or lower. Normalize it toward −16 LUFS; if peaks block the boost, a little Soft clip or compression lets the average level come up without distortion.

Louder than −9 LUFS

Very loud masters get turned down on streaming services, and the squashed dynamics stay. Backing off a few dB often sounds punchier after normalization, not weaker.

Negative stereo correlation

Phase problems make parts of the mix vanish on phones, smart speakers and mono PA systems. Look for a flipped channel, a wrongly polarized microphone, or an over-wide stereo effect.

Long silence at the start or end

Dead air makes tracks feel broken in playlists and wastes listeners' time in podcasts. Use the Trimmer to cut the head and tail, leaving a short natural breath.

Different loudness across a series

Check each episode or track and note its LUFS. Bring them to the same target, then join them in Audio Merger so listeners never need to adjust the volume between parts.

How the measurement works

The file is decoded at its own sample rate whenever the header can be read, so nothing is resampled before measuring. Loudness is computed per channel with the two K-weighting filters from ITU-R BS.1770-4, summed with the standard channel weights, split into 400 ms blocks that overlap by 75%, and gated: blocks below −70 LUFS are dropped, then blocks more than 10 LU under the remaining average. Loudness range uses 3-second windows with a −20 LU relative gate and compares the 10th and 95th percentiles, as in EBU Tech 3342.

Because the whole file is decoded in memory, very long recordings need a lot of RAM — a one-hour stereo file takes roughly 1.3 GB. For multi-hour files, trim a representative section first. Results describe the audio your browser decodes, so a few tenths of a dB of difference from a hardware meter, or from another decoder's handling of MP3 padding, is normal.

Frequently asked questions

Is the LUFS reading broadcast-accurate?

It follows ITU-R BS.1770-4 K-weighting with the EBU R128 gating rules, so it normally lands within a few tenths of a LU of a certified meter. It is still a browser measurement of the decoded file, so for strict delivery specs confirm the final master with a certified meter.

Does analysis upload my file?

No. Decoding and measurement run in your browser with the Web Audio API. The file never leaves your device.

What is the difference between peak and true peak?

Peak is the highest stored sample. True peak estimates the highest level of the reconstructed waveform between samples, which can be higher. Lossy encoders and DACs work on that reconstructed waveform, so true peak is the safer number for clipping.

What should I do if the file is too quiet?

Open Volume Booster and use Normalize or raise the gain. The report tells you how many dB you can add before true peak passes −1 dBTP; if you need more than that, enable Soft clip or compress the dynamics first.

Why is my LUFS value so different from my RMS value?

RMS treats every frequency equally, while LUFS weights frequencies the way the ear hears them and ignores silent passages. A bass-heavy track can read much higher in RMS than in LUFS, and a track with long pauses reads higher in LUFS than its plain average suggests.

Why does the report say my file is clipped when it peaks below 0 dB?

Clipping is detected from runs of consecutive full-scale samples (−0.01 dBFS or above), which means the waveform tops were flattened. A lossy re-encode can lower the peak slightly while keeping the flat tops, so the damage remains audible.

Can it tell me the bitrate or whether an MP3 was upscaled?

No. This tool measures levels and signal health, not codec quality. It cannot reveal bitrate, lossy origin or low-pass cut-offs.

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