Resampling quality compared
Introduction
The
most important instrument for many musicians is the sampler, since it's able to
transform into any other natural or synthesizer instrument. Its core function is
to convert recorded
Samples to the pitch according to the keys you press and then to output them all
together at the sample rate demanded by the connected output device (host
sequencer or soundcard). This process is called "resampling" or "interpolation".
It takes place independently and in real-time for every audible voice. The
applied resampling algorithm substantially decides about the audible playback
quality of a sampler.
The problem:
aliasing
Aliasing occurs whenever an analog waveform needs to be digitized or if a digital signal needs to be converted
(resampled) from one digital resolution to another. And this is exactly what
happens in the sampler when you press a key. You probably know a popular example of
an aliasing error from the apparent backward rotation of wagon wheels in western movies.
It's demonstrated at the next page
Aliasing
examples.
The consequences of aliasing in audio data:
- Loss of harmonics: The added aliasing frequencies destroy
the harmonic relations of a sound, it becomes atonal. Aliasing doesn't know musical harmonic rules.
The added frequencies are just
mathematically related to the digital resolution (sampling frequency)
- Loss of SNR: The added aliasing frequencies
degrade the effective SNR (signal to noise ratio) of
your samples.
- Irreversible: Aliasing frequencies can't be removed subsequently, once
the resampling process produced them. It's as impossible as subsequently telling whether e.g. the number
8 was the result of 3+5 or 8+0.
-
Harsh
"digital" sound: The
audible grade of aliasing depends on the type of the sound and the frequency
relations. In most situations you won't hear the added aliasing frequencies
itself, but what you hear is the general result of aliasing: a rather harsh sound, often referred to as
"digital cold". The picture at the right demonstrates this effect for
graphics: the 1st ellipse is anti-aliased, the.2nd isn't.
The challenge: better resampling
The challenge is to separate "good" from "bad" frequencies.
The sampler needs to remove aliasing frequencies as good as possible while
retaining the source frequencies as good as possible. During the resampling
process this is done by a filter, which removes frequencies outside of the possible target frequency range.
That's not closer to the truth,
but less annoying to the ear then adding them at a completely wrong place. Classical methods
currently found in almost any sampler are linear
interpolation or cubic spline interpolation. Both are fast and in the past years
they were the only possible real-time solution matching the available CPU power. But
both are not ideal for audio
data, they produce heavy aliasing as you can see in some of the screenshots to follow.
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