Normalize

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AviSynth+
Up-to-date documentation: https://avisynthplus.readthedocs.io


Raises (or lowers) the loudest peak of the audio track to a given volume. This process is called audio normalization.
Note that Normalize performs peak normalization (used to prevent audio clipping) and not loudness normalization.

Syntax and Parameters

Normalize(clip clip [, float volume , bool show ] )

clip  clip =
Source clip. Supported audio sample types: 16-bit integer and 32-bit floating-point.
Other sample types (8-, 24- and 32-bit integer) are automatically converted to floating-point.
float  volume = 1.0
Set the amplitude of the loudest audio. Default = 1.0 for peaking at 0dB: for floating-point samples, this corresponds to the range -1.0 to +1.0, and for 16-bit integer samples, this corresponds to the range -32768 to +32767 – the widest range possible without clipping.
  • For a particular peak decibel level, use the equation volume = 10 dB / 20
  • For example, set a -3dB peak with volume = 10-3/20 or 0.7079.
  • Where multiple audio channels are present, all channel gains are set in proportion. For example, if the loudest peak on the loudest channel comes to -10dB, by default a gain of +10dB is applied to all channels.
bool  show = false
If true, a text overlay (see image below) will show the calculated amplification factor and the frame number of the loudest peak.

Normalization and Floating-point Audio

The idea of digital clipping (when the signal is outside the range that can be stored accurately) really applies only to integer sample types; floating-point samples will never become clipped in practice, as the maximum value is around 3.4×1038 – some 29 orders of magnitude (580 dB) larger than 16-bit samples can store.

Normalize is therefore not needed for floating-point audio, but using it is recommended before converting to an integer type, especially if any processing has been done – such as amplification, mixing or equalization – which may expand the audio peaks beyond the integer clipping range.

Examples

  • Normalize signal to 98%
video = AviSource("video.avi")
audio = WavSource("audio.wav").Normalize(0.98)
return AudioDub(video, audio)
  • Normalize each channel separately (eg for separate language tracks)
video = AviSource("video.avi")
audio = WavSource("audio2ch.wav")
left_ch = GetChannel(audio,1).Normalize
right_ch = GetChannel(audio,2).Normalize
return AudioDub(video, MergeChannels(left_ch, right_ch))
LoadPlugin(p + "Waveform\waveform.dll") 
V=BlankClip(pixel_type="YV12", width=480, height=360).Loop
A=WavSource("music.wav")
AudioDub(V, A).AudioTrim(0.0, A.AudioDuration)
ScriptClip(Last, 
\ """Subtitle(Last, "frame "+String(current_frame), align=5)""")
Normalize(volume=1.0, show=true)
Histogram(mode="audiolevels")
Waveform(window=3)
return Last
NormalizeEx2 v1,0.png
(showing frame 2744 where the loudest peak was detected, but note that Amplify Factor is the same for all frames)
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