Free Petite DSP Beta 0.6
Petite DSP is our home written digital signal
processor. We developed it to use with an electric guitar and other instruments,
but it can perfectly be used while playing cd's or tapes.
In Beta release 0.6 are the following effects
You can download the executable
System requirements: Pentium 166+ for 44k1 stereo
dsp, (although you can try a 486 at 22 Khz mono), full-duplex 16 bits soundcard,
The program is tested using equipment like a
PR200Mx, Pentium 133, Pentium 200, using a OPTI924, an Aztech, an SB AWE
64, and a few 20 Euro cards ('addonics' trident pci,.crystal cmi). It all
seems to work fine.
Your sound card has to work full duplex. Some
direct-x drivers prevent your sound card from doing so, so please check,
sometimes the option 'direct-x / full duplex' can be changed. Check -Configuration
Screen - Sound&Game controllers - 'YourSoundCard' Wave Audio - properties
- ... .
Depending on your hardware you can increase or
decrease buffer sizes, nr buffers used and your numbers of pre filled buffers.
Especially, if you want to use this dsp real-time,
try to reduce your buffer size and use 44k1 stereo.
The latency can be about 40-70ms up to 300 ms.
Normal settings are:
Buffer Size:32-1024 byte;
Prefill- bufs: 50-350; (increase prefill as buffersize
decreases and vice versa);
Number of buffers (200-1000);
I've given full control over these hardware settings
to make it run optimal on different systems.
One can not say which settings are 'best' - try,
because it's rather safe. Look at the presets to understand the relations
(especial memory consumption). Don't make your prefill bigger than about
half the amount of buffers used.
Please be aware that this program can take up
to 100% processor time, and always should be terminated normally, which
as far as I know, it always will.
I disabled the save-to-file function in this
version, since this was rather system recource consuming and therefore
a bit tricky.
A Phaser, Flanger and a chorus all use the same
A delayed signal is added to the original signal.
The delay itself changes in time, with a Low Frequency Oscillator (LFO).
The shape of the LFO has also influence,
sinus and exponential
shapes are common used.
For the chorus/flanger I used a triangle LFO
shape. For the phaser an exponential LFO shape (typical for a phaser).
A Chorus adds a vibrating -almost- echo to the
signal (the delay is too short for your ears to hear it as a delay). So
a voice sounds as if two people are singing, or two people sound like a
chor. Or a guitar is some more 'full' as if there were two.
A phaser uses a very short delay. This way, some
frequency's get filtered out (if the delay is the same as one half wavelength
the frequency is filtered out). Because the delay is changing, it gives
a very 'spacy' sound. Apply it on a drum or on a whole mix.
A flanger is- in my opinion, just weird. But you
may like the sound. By setting the initial delay to almost zero, it will
act a bit as if it were a phaser. With another LFO wave type though.
Nice for guitar effects or 'scrambled' voice.
Depending on the settings you can get different
Typical settings for Phaser, Flanger and Chorus
Please Note that if you reduce the depth of the Effect
to 0, the remaining thing is an ordinary echo.
||0 dB added to input
||< 0.5 Hz
||1/10-1/3 of max.
||-3 - 0 dB
||-10 - -3 dB
A smart kinda fir filter cuts noise and real
high tones out of your input source. It is applied before the signal goes
through other effects. I think the best setting is halfway. (>14K7 cut
off at 44k1, >8K cut off at 22Khz)
All effect signals are added into one master.
Please adjust your master to avoid clipping.
Internal processing is partly float and partly
24-28 bits integer, so don't worry about 16 bits quality.
Note that the original signal is added the same
as the effects. Setting of this has no effect on the source going into
the effects. You have to adjust your windows mixer line/mic in if your
input sound is to loud or to soft. The only effect applied before the input
sound goes through the effects is the noise filter.
I added it for fun, because it's quite simple:
subtracting left from right and vice versa results
in a signal where the equal parts of a stereo signal are filtered out,
so you only hear what typically came out of you left or right speaker originally.
Applied on a mono signal the result simply is
that you hear nothing.
You can adjust the balance (the L/R centre of
the sound filtered out).
Whether Karaoke works or not depends on your
sound source. Just try.
Normally you don't mix the Karaoke signal with
the original input for the karaoke effect, although you can 'boost' left
or right placed instruments by adding karaoke...