AUD · Audio tools

Audio Volume Changer

Change the level of an audio file by a fixed amount in decibels. The gain runs from -24 dB to +24 dB, and the same value applies to every channel so the stereo image does not shift.

Decibels, not percentages

Audio level is measured logarithmically. Each 6 dB step doubles or halves the amplitude of the waveform:

  • +6 dB doubles the amplitude
  • +12 dB quadruples it
  • -6 dB halves it
  • 0 dB leaves the samples untouched

Perceived loudness is a separate scale. Doubling amplitude does not sound twice as loud; a change closer to 10 dB usually does.

Clipping above 0 dBFS

Digital audio has a ceiling. Samples are stored in a range that peaks at 0 dBFS, and anything pushed past it has nowhere to go. On export those samples are clamped flat, which replaces the top of each waveform peak with a straight line and adds harsh upper harmonics.

A track already mastered near 0 dBFS has no headroom, so even +1 dB will clip it. The peak and RMS meters next to the transport show where the level sits while the file plays.

Comparing against the original

The bypass control in the transport swaps between the processed and unprocessed signal at the same playback position. Since a level change is the easiest thing to mistake for an improvement, switching back and forth is the fastest way to judge whether the gain is doing what you want.

Re-encoding the same format

The export comes back in the container the file arrived in, at a bitrate read from the source, so a level change does not quietly turn an MP3 into a WAV or inflate a 96 kbps file. Changing format is the Audio Converter’s job.

That does mean a lossy source goes through its encoder once more, and each pass discards detail that cannot be recovered. Convert to WAV or FLAC before the level change when the file still has several edits ahead of it.

Frequently Asked Questions

It applies a fixed gain, measured in decibels, to every sample in an audio file. It reads WAV, MP3, M4A, AAC, Ogg, Opus, FLAC, AIFF, and WebM audio, and exports in the format the source arrived in.

+6 dB doubles the amplitude of the waveform, but perceived loudness follows a different curve. A change of roughly +10 dB is what most listeners describe as twice as loud.

It can. Samples pushed past 0 dBFS are clamped to the maximum value on export, which flattens the peaks and adds harsh harmonics. Watch the meters during playback and back the gain off if they sit at the top.

Not exactly. Each export re-encodes the audio, and lossy formats such as MP3 and Ogg discard detail every time. Convert the file to WAV or FLAC first when it has to survive several rounds of processing.

No. The gain applies to the whole file, and a range marked on the waveform only sets what looped playback repeats. Cut the section out with the Audio Trimmer first, then change its level.

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