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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.
Drop a track to measure levels, spot clipping and quiet parts, and see what to fix next — all on your device.
Drop an audio file here
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MP3 · WAV · FLAC · AAC · M4A · OGG
Analyzing…
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.
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.
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.
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.
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.
| Destination | Loudness target | True-peak ceiling |
|---|---|---|
| Spotify, YouTube, Tidal | about −14 LUFS | −1 dBTP |
| Apple Music | about −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.
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.
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.
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.
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.
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.
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.
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.
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.
No. Decoding and measurement run in your browser with the Web Audio API. The file never leaves your device.
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.
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.
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.
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.
No. This tool measures levels and signal health, not codec quality. It cannot reveal bitrate, lossy origin or low-pass cut-offs.