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VST (Virtual Studio Technology) is a popular format from Steinberg for loading plug-in effects into audio sequencers, hard disk recorders, sound editors, and trackers   ConvolverVST is a convolution plug-in for (version 2.4 of) that format.


A potentially interesting application of convolverVST would be with the Audiotrak Prodigy 7.1 sound cards, which can filter the input signal in real time without a sequencer.  (I have not tried this.) This will allow convolution processing to be applied to sound from any application running on your system. See the Audiotrak web site for more details.

A similar approach — application of a filter to the sounds generated by any program installed on your computer — is available through the SonicStage Mastering Studio Audio Filter feature available with some Sony Vaio machines.


The convolverVST plug-in is under active development.  The current version has the following limitations:

  • There is no specific midi support
  • There is no offline processing support
  • The presets do not include the filter filename (so using multiple versions of convolverVST may cause confusion)
  • The accumulating process mode is not supported. (It just calls the replacing processing mode.)

The VST standard has evolved over time and is not precisely specified.  Moreover, different hosts have different requirements of their VST and use the interface in different ways.  I would very much welcome feedback on what works and what doesn't.

For example, I have been able to get the current version of convolverVST to work with

Getting started

To install convolverVST:

1. download convolver and unzip it and run Setup.exe.

2. convolverVST requires libsndfile and libfftw DLLs to run.  For many host applications, these must be in the execution path.  To add the them to the execution path, right click on My Computer in Windows Explorer, select Properties, the Advanced tab, and the Environment Variables button and append ;C:\Program Files\Convolver\Convolver (or the path to which you installed Convolver in the first step) to either the User or System Path variable using the corresponding Edit button:

3. Now start your host and scan for new VST plug-ins using your host application. (Some host applications require plug-ins to be dragged and dropped on them.)

4. Depending on your host, either at the scanning stage, or when you then load ConvolverVST, you will get a dialogue box reminding you that no filter has yet been selected.  Depending on your host, this may happen more than once.

5. Now get convolverVST's property page up.  How you do so depends on your host:

Hit the Filter/config button and select a WAV (or other supported sound file format) file
The file will be convolved channel by channel with the source.  The file must have the same number of input and output channels and sample rate as the source that you are playing back and the number of speakers that you are using.  Raw PCM (.PCM) files are assumed to be single-channel 44.1kHz 32-bit float files.  Similarly .DBL files are assumed to be single-channel 44.1kHz 64-bit float files.  If you need to specify a different sample rate with a raw file, use a config file.
This is not the best approach theoretically for room correction, but is simple and may give you good sound anyway; or
a configuration (text) file
This is more complex, but more flexible approach.  There are also some examples, although they will not be intelligible without looking at the config file instructions first.
The links page has some pointers to sources of impulse responses that you can try, just to get going. See duffroomcorrection if you want to record your own room impulse responses.

6. The input channels are also scaled by 10 Atten'n / 20. So negative values will help to to prevent clipping and positive ones will result in amplification. If you hear no playback, try raising this value. The negative of the Estimated gain value shown in the filter description provides a good starting point. The (negative of the) Peak gain value will guarantee that there will be no overflows, but is extremely conservative. If you push the shift key while dragging the slider, you will get finer control.

7. The Partititions setting allows you to tune the convolution algorithm to your cpu (and the size of its cache).  Try values such as 0, 2, 4, 8 or 16.  A value of 0 will use straight overlap-save convolution. (A setting of 1 will effectively do the same, or more precisely overlap-add, but slightly less efficiently.)  Also, the larger the number of partitions, the shorter the lag between input and output.

8. Convolver tunes itself for a particular filter and number of partitions when the filter is loaded or the number of partitions changed.  The Tuning rigour setting allows you to specify how much effort the convolution routines put into tuning themselves to your system:

specifies that, instead of actual measurements of different algorithms, a simple heuristic is used to pick a (probably sub-optimal) method quickly.
finds an optimized method by actually and measuring the execution time of candidate methods, so there may be a pause while this happens. Depending on your machine, this can take some time (often a few seconds). This is a good default option.
is like Measure, but considers a wider range of algorithms and often produces a “more optimal” method (especially for large convolutions), but at the expense of much longer tuning time (minutes). This option also considers multi-threaded execution, which may be a benefit for longer filters and multiple processors (or even multi-core and perhaps even hyperthreading processors).
is like Patient, but considers an even wider range of algorithms, including many that are unlikely to be fast, to produce the most optimal calculation method but with a substantially increased tuning time.
Take 1 minute
Finds the best method that it can within a minute for each filter path.

The tuning results are retained for reuse in a file called wisdom.fftw.

9.You are now ready to use convolverVST...

Technical features

  • convolverVST is built with version 2.4 of the VST SDK and the CVS version of VST GUI
  • 32- and 64-bit samples are accepted, but internal processing is 32-bit
  • Speaker arrangement is supported (kSpeakerArrMono, with each speaker kSpeakerM (mono))
  • VstPinProperties are supported
  • Initial delay is set, reflecting the latency of the current filter path
  • VstParameterProperties are supported


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