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ck.altostratus
Description
Ck.altostratus: a strange midi instrument based on PeRColate īs blotar: PeRColate is an open-source distribution of a variety of synthesis and signal processing algorithms for Max, MSP, and Nato. It is centered around a (partial) port of the Synthesis Toolkit (STK) by Perry Cook (Princeton) and Gary Scavone (Stanford CCRMA). Like the STK, it provides a fairly easy to use library of synthesis and signal processing functions (in C) that can be wired together to create conventional and unusual instruments. Also like the STK, it includes a variety of precompiled synthesis objects, including physical modeling, modal, and PhISM class instruments; the code for these instruments can serve as foundations for creating new instruments (one example, the blotar, is included) and can be used to teach elementary and advanced synthesis techniques. Given it's STK heritage and educational function, PeRColate is largely un-optimized, though all the objects run on a 80MHz 7100, which is pretty good. PeRColate also includes a number of objects not from the STK; some are from RTcmix and others are our own evil creations, designed to crash your computer, but only after making some kind of interesting sound or image.
free vst plugin, pc, mac
free vst plugin, pc, mac
Category
Date
Jan 9, 2007
Contact name
Email
Link ID
17056
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Other links at Free VST plugins > other
Normalizer denormals eliminator:
denormalisation is an issue with some audio hosts that don't switch off the plugin's audio task when the song is stopped or a track contains silence (or with instruments: no key is pressed while a sound decays).
The problem happens when the CPU (precisely: the floating point unit (FPU) inside of it) detects extremely small numbers and wishes to process them with the same precision as usual. Then it switches into 'denormal' mode which is another way of representing small floating point numbers with the available bit range.
Converting between the two states takes a lot of time for the FPU and thus might cause heavy load jumps.
Some algorithm such as delay lines and filters are prone to denormalisation as they work with feedback structures.
Here, if an impulse is followed by absolute silence, the resulting signal gets smaller and smaller but in theory, it can never reach zero.
In practice, there is an internal threshold of the FPU that switches to denormal mode to carry on processing those small numbers (which might not be of any use for proper operation).
Currently, the Pentium 4 processor seems highly addicted to denormalisation. It turned out that its threshold is actually very much higher than of former Pentium types.
Plugins that used to work fine with older CPUs might slow down the entire system when used with a P4.
But other FPUs might as well suffer from denormalisation issues.
Most developers are aware of denormalisation although the methods to avoid it are many. Some algorithms ask for the current FPU status from time to time, reacting to it by changing small values to zero.
Other developers prefer adding an an extra signal to the audio which is carried along the entire process of the plugin, thereby forcing it to remain above the internal threshold.
Should the input become silent, the additional noise will still keep the whole thing from denormalizing.
All the digitalfishphones plugins actually work this way, and so does the normalizer.
Normalizer usage:
Fill the first insert slot with this plugin. Put the plugin that's supposed to cause denormalisation into the second slot.
The normalizer plugin lets you choose between two methods: adding AC or DC signal.
Here, AC is a square wave ringing at the nyquist frequency (half the sampling rate, e.g. 22.05kHz).
Choose this method whenever the following plugin does not apply intensive low-pass filtering, thereby eleminating our injection noise.
In that case, the DC method could be used. It just adds a static offset to the signal. This DC level (when set pretty low) should be no issue with further processing of the resulting output.
No matter which method you choose from, the noise floor should be lower than the lowest value that your D/A converter can handle.
And in fact, a level of about -300 to -200dB is usually enough. Sometimes you can even go much lower.
You will have to experiment with the proper level because the internal working of the faulty plugin is most certainly unknown to us.
The more stages it involves, the more complex the whole thing is.
free vst plugin. pc, mac
denormalisation is an issue with some audio hosts that don't switch off the plugin's audio task when the song is stopped or a track contains silence (or with instruments: no key is pressed while a sound decays).
The problem happens when the CPU (precisely: the floating point unit (FPU) inside of it) detects extremely small numbers and wishes to process them with the same precision as usual. Then it switches into 'denormal' mode which is another way of representing small floating point numbers with the available bit range.
Converting between the two states takes a lot of time for the FPU and thus might cause heavy load jumps.
Some algorithm such as delay lines and filters are prone to denormalisation as they work with feedback structures.
Here, if an impulse is followed by absolute silence, the resulting signal gets smaller and smaller but in theory, it can never reach zero.
In practice, there is an internal threshold of the FPU that switches to denormal mode to carry on processing those small numbers (which might not be of any use for proper operation).
Currently, the Pentium 4 processor seems highly addicted to denormalisation. It turned out that its threshold is actually very much higher than of former Pentium types.
Plugins that used to work fine with older CPUs might slow down the entire system when used with a P4.
But other FPUs might as well suffer from denormalisation issues.
Most developers are aware of denormalisation although the methods to avoid it are many. Some algorithms ask for the current FPU status from time to time, reacting to it by changing small values to zero.
Other developers prefer adding an an extra signal to the audio which is carried along the entire process of the plugin, thereby forcing it to remain above the internal threshold.
Should the input become silent, the additional noise will still keep the whole thing from denormalizing.
All the digitalfishphones plugins actually work this way, and so does the normalizer.
Normalizer usage:
Fill the first insert slot with this plugin. Put the plugin that's supposed to cause denormalisation into the second slot.
The normalizer plugin lets you choose between two methods: adding AC or DC signal.
Here, AC is a square wave ringing at the nyquist frequency (half the sampling rate, e.g. 22.05kHz).
Choose this method whenever the following plugin does not apply intensive low-pass filtering, thereby eleminating our injection noise.
In that case, the DC method could be used. It just adds a static offset to the signal. This DC level (when set pretty low) should be no issue with further processing of the resulting output.
No matter which method you choose from, the noise floor should be lower than the lowest value that your D/A converter can handle.
And in fact, a level of about -300 to -200dB is usually enough. Sometimes you can even go much lower.
You will have to experiment with the proper level because the internal working of the faulty plugin is most certainly unknown to us.
The more stages it involves, the more complex the whole thing is.
free vst plugin. pc, mac
Category:
2. tuna
Tuna is a guitar tuner ...
Guitar tuner made with Max/MSP.
Pluggo is an extraordinary set of multiplatform audio plugins. (AU/RTAS/VST) I'm sharing several free ones made with MAX/MSP . You must have Pluggo, Pluggo Demo, or a free Pluggo Runtime installed to use it. Installation will create a Pluggo folder inside /Library/Audio/Plug-ins/VST (OSX) - or - /Digidesign/ProTools (OS9). To install my pluggos - simply drop it into Pluggo folder and start your Cubase/Logic/ProTools.
free vst plugin, mac.
Guitar tuner made with Max/MSP.
Pluggo is an extraordinary set of multiplatform audio plugins. (AU/RTAS/VST) I'm sharing several free ones made with MAX/MSP . You must have Pluggo, Pluggo Demo, or a free Pluggo Runtime installed to use it. Installation will create a Pluggo folder inside /Library/Audio/Plug-ins/VST (OSX) - or - /Digidesign/ProTools (OS9). To install my pluggos - simply drop it into Pluggo folder and start your Cubase/Logic/ProTools.
free vst plugin, mac.
Category:
Voxengo Latency Delay:
Latency Delay is an auxiliary latency compensation plug-in which allows you to compensate latency introduced by modular real-time VST plug-ins and instruments which introduce latency but do not try to report it to the host. Latency Delay introduces 10000 samples latency itself and delays the audio signal by [10000 minus specified sample value] samples. Please note that host audio application should support latency compensation. This plug-in is only a helper and should be used together with other plug-ins.
Latency Delay is an auxiliary latency compensation plug-in which allows you to compensate latency introduced by modular real-time VST plug-ins and instruments which introduce latency but do not try to report it to the host. Latency Delay introduces 10000 samples latency itself and delays the audio signal by [10000 minus specified sample value] samples. Please note that host audio application should support latency compensation. This plug-in is only a helper and should be used together with other plug-ins.
Category:
4. vst-chess
VST -Chess is the world's first chess game
for the VST plugin standard.
You have many configuration options,
different tile sets are included and you can
play against another (human) player or
against the computer.
free vst plugin. pc
for the VST plugin standard.
You have many configuration options,
different tile sets are included and you can
play against another (human) player or
against the computer.
free vst plugin. pc
Category:
5. ggain
GGain is a simple volume knob. It boosts or cuts the level of the input signal by up to 12dB.
Parameters
Gain: The gain applied to the incoming signal.
free vst plugin, pc.
Parameters
Gain: The gain applied to the incoming signal.
free vst plugin, pc.
Category:
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