Audio Equalizer
Apply a disclosed three-band peaking equalizer locally, prevent PCM clipping with one uniform safety scale, and download a complete PCM16 WAV.
Privacy: your files never leave your device. All processing happens locally in your browser.
How to use
- 1.Choose one browser-decodable audio file within the 50 MiB and five-minute decoded limits.
- 2.Set Bass, Mid, and Treble to whole values from -12 through +12 dB, then apply the equalizer locally.
- 3.Check the raw peak, output peak, safety scale, sample rate, and frame count; preview and download the PCM16 WAV.
About Audio Equalizer
Audio Equalizer applies three disclosed peaking filters to audio that your browser can decode. Choose a file, set Bass, Mid, and Treble from -12 through +12 dB, and create a new PCM16 WAV without uploading the source. Bass is centered at 100 Hz, Mid at 1,000 Hz, and Treble at 10,000 Hz when the decoded sample rate permits it. At a low sample rate, the treble center is reduced to 45 percent of that sample rate so it stays below the Nyquist limit instead of asking an invalid digital filter to operate outside the available spectrum.
Each band uses a Q value of 1 and the peaking-EQ biquad equations published in the W3C Audio EQ Cookbook. The page converts the selected decibel gain into the coefficient parameter defined by that note, normalizes every coefficient by a0, and runs the resulting second-order recursive filter over each decoded channel. The three sections are cascaded in Bass, Mid, then Treble order. This is a compact three-band equalizer, not a graphic 10-band system, a linear-phase mastering processor, an automatic room correction system, or a promise that one setting improves every recording.
Zero on all three controls is a genuine flat path. In that case, the tool copies the decoded channel samples without running a nonzero filter section. Positive values boost energy around a band center, and negative values cut it. A peaking filter affects a region around its center rather than every frequency below or above a hard boundary. Q=1 is a disclosed product choice that provides a broad adjustment; it is not inferred from the source file and cannot be changed on this page.
Cascaded boosts can create samples above digital full scale even when the input did not clip. Before WAV encoding, the tool measures the absolute peak across every processed sample and every channel. If that raw peak exceeds 0.99, one uniform scale factor reduces the entire processed result so the largest magnitude becomes 0.99. The interface reports the raw peak, output peak, and exact safety scale. This prevents the PCM encoder from independently chopping individual peaks, but it can reduce the overall level. It is not loudness normalization and does not target LUFS, RMS, replay gain, or a streaming-platform specification.
Processing remains in the current browser tab. Web Audio decodes the selected MP3, WAV, M4A, AAC, Ogg, WebM, or FLAC when the current browser and operating system support the actual codec. A familiar extension is not a guarantee: containers can hold codecs that a browser cannot read. A decode failure produces an error and no stale download is presented. Lizely does not receive the audio bytes, decoded samples, EQ settings, filename, or result.
Limits protect the tab from unexpectedly large decoded data. Input is capped at 50 MiB. Decoded audio must be five minutes or shorter, have one through eight channels, use a sample rate from 8,000 through 192,000 Hz, and contain no more than 30 million channel samples. The tool rejects an over-limit file before filtering and never silently truncates the end. Each channel keeps the same number of sample frames, and all accepted channels are filtered independently with identical settings.
The download is a new uncompressed PCM16 WAV. It does not keep the original MP3 or AAC bitrate, compression mode, tags, artwork, chapters, loop markers, or container metadata. Web Audio may resample while decoding, so the result reports and writes the decoded sample rate rather than claiming the source container rate survived unchanged. Float samples are converted to signed 16-bit little-endian PCM and interleaved by frame. The WAV header records the actual channel count, sample rate, byte rate, block alignment, bit depth, and complete data length.
Use the preview and reported numbers as checks, not as proof of a desirable mix. Listen on the speakers or headphones that matter, compare against the original at a similar perceived level, and avoid judging only from louder output. Strong boosts may emphasize noise, sibilance, rumble, or existing distortion. A cut is often safer than stacking several boosts. Use a full audio editor when you need precise center frequencies, adjustable Q, spectrum analysis, automation, linear-phase processing, loudness metering, dithering controls, codec selection, or metadata preservation.
The verification contract is numeric. Tests cover eight external decibel-to-amplitude values from -12 through +12 dB, coefficient identity at 0 dB, an unchanged flat channel, a boosted impulse, uniform 0.99 peak safety, invalid gains, mismatched channels, and safe filenames. Production calls those same coefficient, filtering, peak, and filename functions. The final WAV uses the already tested shared PCM16 encoder, and browser tests read its RIFF fields and sample data rather than accepting a visible success message alone.
Methodology & sources
Decode one bounded local file with Web Audio, reject invalid duration/channel/rate/sample budgets, and apply three cascaded W3C Audio EQ Cookbook peaking biquads with Q=1. Centers are 100 Hz, 1,000 Hz, and min(10,000 Hz, floor(sampleRate*0.45)); gains are integer -12 through +12 dB. Measure the raw peak across all processed channels, uniformly scale to a 0.99 maximum only when needed, and encode the complete equal-length channels as tested interleaved little-endian PCM16 WAV. Report centers, frames, peaks, safety scale, and bytes; revoke stale Blob URLs.
Frequently asked questions
- What frequencies do the three controls use?
- Bass is 100 Hz, Mid is 1,000 Hz, and Treble is normally 10,000 Hz. For low sample rates, Treble becomes 45% of the decoded rate so it remains below Nyquist.
- Why can the safety scale be below 1?
- Combined boosts can exceed digital full scale. When raw peak is above 0.99, the tool scales all processed channels uniformly so the output peak is 0.99 rather than clipping individual samples.
- Does the download keep my original codec and tags?
- No. The result is a new PCM16 WAV. Original compression, bitrate, artwork, tags, chapters, and container-specific metadata are not preserved.
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