Skip to content
Lizely

Volume Changer

Raise, reduce, or mute a browser-decodable audio file locally and download a complete PCM16 WAV result.

Privacy: your files never leave your device. All processing happens locally in your browser.

How to use

  1. 1.Choose a supported audio file no larger than 50 MiB and wait for the file name to appear.
  2. 2.Set linear output volume from 0% through 400%, then select Change volume.
  3. 3.Preview the WAV, inspect its peak and clipped-sample count, and download only after listening to the complete result.

About Volume Changer

Volume Changer adjusts the sample amplitude of an audio file entirely in the current browser tab. Choose a browser-decodable MP3, WAV, M4A, AAC, Ogg, WebM, or FLAC file, select a volume from 0% through 400%, and export the complete result as a 16-bit PCM WAV file. The source audio is not uploaded, queued, analyzed by an external service, or retained by Lizely.

The percentage is a direct linear amplitude multiplier. At 100%, samples keep their decoded amplitude. At 50%, every sample is multiplied by 0.5. At 200%, every sample is multiplied by 2. Zero percent writes silence while preserving the decoded duration, sample rate, and channel count. This is not a decibel control: doubling linear amplitude is approximately a 6.02 dB change, not a 100 dB change.

The browser’s Web Audio decoder first converts the complete encoded file into floating-point channel samples. Format support therefore depends on the browser, operating system, and codec build. A filename extension is not proof that the media can be decoded. If decoding fails, the tool stops and shows an error rather than creating an empty or partial output.

Each finite decoded sample is multiplied by the selected gain. Values above +1 or below -1 are clipped to the supported full-scale range before PCM encoding. The result reports how many samples crossed that boundary and shows the peak after clipping. A nonzero clipped-sample count is evidence that the gain may have introduced audible distortion, although zero clipping does not guarantee ideal perceived loudness.

The exported WAVE file uses a RIFF container, little-endian integer fields, format code 1 for uncompressed PCM, the original decoded sample rate, the original number of channels, and 16 bits per sample. Channel samples are interleaved frame by frame. Negative full scale maps to -32768 and positive full scale maps to 32767, matching the asymmetric range of signed 16-bit audio.

Output is always WAV even when the source is compressed. WAV is broadly inspectable and avoids adding a new codec or encoder dependency, but it is usually much larger than MP3, AAC, or Ogg. The tool does not preserve source tags, album art, chapters, loop points, cue markers, encoder metadata, or the original compression settings. The downloaded filename ends in -volume-adjusted.wav.

The input file is limited to 50 MiB. Decoded duration is limited to five minutes, channel count to eight, and total decoded channel samples to 30 million. These limits bound memory allocation and main-thread work on ordinary laptops and mobile devices. A compressed file can expand substantially after decoding, so the decoded limits matter even when the selected file is small.

Changing the file or percentage invalidates the previous result and revokes its temporary object URL. A job identifier prevents an older asynchronous decode from overwriting newer settings. The AudioContext is closed after processing. The generated download remains temporary and disappears when the tab closes or when a new operation replaces it.

Linear gain is not loudness normalization, mastering, limiting, compression, noise reduction, or broadcast compliance. Perceived loudness depends on frequency balance, duration, dynamics, speakers, and listening conditions. This tool does not calculate LUFS, true peak, replay gain, or platform-specific targets. For professional delivery, use a measured audio workflow and monitor the exported file.

For a safe edit, begin near 100%, preview the full output through suitable headphones or speakers, and check the clipped-sample report. Reduce the percentage if distortion appears. Keep the original file because the operation cannot restore information already clipped in the source and repeated PCM conversions can complicate version control. Verify the downloaded duration and channels before replacing any production audio.

Methodology & sources

Validate file type and a 50 MiB bound; decode the complete file with Web Audio; reject durations above five minutes, more than eight channels, or more than 30 million channel samples; multiply every finite sample by the selected 0–4 linear gain; clamp to digital full scale while counting crossings; encode all channels as interleaved little-endian RIFF/WAVE format-code-1 PCM16 at the decoded sample rate; revoke stale object URLs and close the decoder context.

Frequently asked questions

Is 200% volume the same as adding 200 decibels?
No. It doubles linear sample amplitude, which is approximately a 6.02 dB amplitude increase before clipping.
Why is the output a WAV file?
PCM16 WAV can be encoded locally without a new lossy codec dependency and exposes a deterministic complete result.
What does clipped samples mean?
It counts multiplied samples that exceeded digital full scale and had to be limited to -1 or +1.
Is my audio uploaded?
No. File decoding, gain, WAV encoding, preview, and download all occur in the current browser tab.

Related tools