DSPs can be used to add extra playback features, some of
which can help improve audio quality, or at least minimize
artefacting. DSPs need to be listed in a specific order for
maximum effectiveness. That being as follows:
Though most of the Available DSPs are
fairly self-explanatory, I’ll describe the more relevant
This enables the use of the Equalizer displayed in
the Equalizer tab.
By default no presets are
included. You can however find a
bundle here (extract the contained folder to the
Foobar directory), which can be loaded via the Load
resets all bands to 0dB. Store Preset can be
used to save custom equalizer settings if you decide to
create/edit any. Auto level adjusts any equalizer
adjustments so that no positive amplification is applied to
any of the frequency bands, i.e. 0dB is the “highest”
value (This may help reduce possible distortion without
changing the “shape” of the frequency band), e.g. in the
image above +5dB is the highest level of
amplification, accordingly all frequency bands are
lowered by -5dB (-8dB change to -13dB
and so on).
This enables the resampling of audio sources to a
specified sampling rate. Many soundcards do this
automatically already when playing sources with a certain
sampling rate, e.g. 44.1 kHz resampled to 48 kHz. Depending
on the Soundcard this resampling in hardware can introduce
audio artefacting. The resampler which Foobar includes is of
better quality, and provides better resampling than many
soundcards will (albeit with increased CPU usage).
If you use an AC97 Soundcard/Integrated
Audio, any Creative soundcards or any of the nForce APUs,
then it’s recommended to use this (in theory). I would still
recommend testing both and ensuring you are actually getting
some benefit otherwise you’re just wasting processing power.
Conversely, soundcards without such
Envy24 based), there’s no need to use this DSP. Now
select the Resampler tab.
Target sample rate.
This specifies the sampling rate for audio sources to be
resampled to. 48000Hz as a result, will be the
preferable choice from the drop-down menu. Lower and higher
rates are also available, though can be ignored unless you
have a specific reason to select them (higher will
not lead to better sound quality). Again, the main purpose
is to avoid audio artefacting caused by poor resampling by
the soundcard or integrated audio.
Crossfeed works by simulating the audio absorption
characteristics of the head, mixing a delayed section of the
right channel with the left channel and vice versa. This
provides a more natural listening experience for those using
Headphones. If you’re listening to Speakers then this occurs
naturally and there’s no need to use this DSP.
This enables the use of the volume slider in Foobar,
otherwise the standard Windows Mixer controls will control
the volume level (depending on the Output method
selected – covered later - this may prove to be the only way
to control playback volume in Windows).
Soft clipping limiter.
This applies a 6dB limit to audio playback, which will
likely introduce distortion, as opposed to clipping. As
foodev on the
Hydrogenaudio Forums said, this “can audibly modify
the sound indeed, but it completely eliminates any clipping.
Also, if your volume is set to -6dB or lower, [soft
clipping] limiter triggers only in places where samples
would normally get truncated. Conclusion? Set wave volume to
max in windows volume control, & use fb2k's internal volume
control instead, it eliminates clipping most of the time, &
makes [soft clipping] limiter never (or hardly ever) trigger
if enabled. Maybe there are different points of view on this
matter, but I find [soft clipping] limiter less annoying
than clipping or even auto-attenuation”.
This DSP uses read-ahead on the source and scales down the
audio signal when it detects clipping will occur resulting
in minimal, if any, distortion during playback.