Tom's Audio Processing plugins
for audio engineering on the Linux platform
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General Information


Should be pretty easy. Download and untar the archive and cd into the resulting directory tap-plugins-x.y.z. If you store your plugin .so files in a different directory than /usr/local/lib/ladspa (which is the default) then please edit the top of the Makefile first of all. There is a default place also for RDF metadata. Set it according to your own needs. Because the plugins do not require any special library (apart from the standard GNU C library), there is no ./configure script. All you have to do is issue a make and (as root) a make install. If you don't get any errors (you shouldn't), fire up your host and check out your new plugins. You may also want to check the README file for additional information.

A few words about signal routing

When connecting plugins to audio inputs/outputs and to each other in order to build a digital audio processing chain, it is important to understand how audio is actually routed between individual input/output ports. Plugins can have a different number of audio inputs and outputs, and routing is handled by the host.

The TAP-plugins have either 1 input and 1 output (mono plugins) or 2 inputs and 2 outputs (stereo plugins). No other combinations are used, although it would be perfectly legal for a plugin to have 1 input and 2 outputs, for example.

When you apply a mono plugin to a mono track, everything is clear. The one and only input of the plugin gets connected to the mono signal, and the single plugin output is routed back somewhere (to the input of another plugin, or to a master out, etc). Another simple case is when you apply a stereo (2 in / 2 out) plugin to a stereo track. Here again, everything gets connected as you would expect.

But what happens if you apply a mono plugin to a stereo track (or put it in a chain of plugins after a plugin with stereo outputs)? In this case, actually two instances are created of the same plugin, and the two input channels are processed by these two separate plugins to get two output channels. The two instances are binded to the same GUI controls, so you see and adjust only one plugin GUI. But keep in mind that it's actually two plugins running in the background, consuming twice as much memory and CPU power as one single plugin would cost. The CPU usage metrics (provided as part of the plugin manuals) for mono plugins assume you are actually using only one instance, so you should double the figures when applying them to a stereo track.

The last possible arrangement is when you apply a stereo plugin to a mono track (or put it in a chain of plugins after a plugin with only one output). Here the stereo plugin receives the same data on both input channels. This can be an important issue because there are some effects requiring that audio is received on only one input and the other input muted (i.e. feeded with zeroes) if the input is actually mono. (See the Haas effect switch of the TAP Stereo Echo plugin for an example).

Given the above routing behaviour, TAP-plugins have been written in a manner that minimizes the number of stereo plugins. If the stereo version of a plugin would process two input signals identically and separately, then there is no stereo version of the plugin since applying the mono plugin to a stereo track yields the exact same effect. A plugin is implemented as stereo if and only if it uses an algorithm that results in differences between the output channels even with identical input data (echoing and reverberation algorithms are an example of this). If one of the input channels needs to be muted to achieve a particular effect when the inputs are identical, there is a user controllable toggle button built in the plugin to achieve this.

About CPU usage metrics

CPU usage metrics are provided for each plugin. This is not a very accurate way to tell how much a plugin setup costs, but it should provide a general idea about how expensive each plugin is, and it is better than nothing. Please don't rely on these figures when you need to predict the system load in a situation where the results of system overload (stopping of the transport, falling out of sync, etc) are unacceptable. For example, if you are recording a live show or doing live PA work, never ever go into calculations that result in saying that you will need only 98% of your CPU power when using such-and-such plugins. This is extremely dangerous, and there are much easier ways to commit suicide as well.

The CPU usage figures were measured running the plugins within Ardour, and recording the increase of "DSP Load" indicated at the top of the Ardour Editor window. The machine used for measurements was a uniprocessor Pentium IV machine with a processing power of approximately 3395 BogoMIPS. Mono plugins were applied to mono tracks, and stereo plugins were applied to stereo tracks (read the above section about routing to understand why this is important). Measurements were made at both 44.1 kHz and 96 kHz sampling frequencies. CPU usage is proportional to the sampling rate (at least in theory), so at 48 kHz you should expect somewhat greater CPU demand than at 44.1 kHz, and at 192 kHz expect roughly twice as much CPU usage as the value at 96 kHz. The actual figures are provided as part of the plugin manuals.

About latency issues

To be able to perform complex tasks, several plugins have some latency. The value of latency means the number of samples the output is "late" when compared to the input. You can tell if a plugin has latency, because it is always indicated in the "Summary of user controls" table of the plugin manual. Certain plugins have a latency of fixed samples, others have a fixed time of latency, which yields different latency values at different sample rates. And there are plugins that have latency values dependent on user settings.

A latency value of a few hundred samples is not a long time (when you are waiting for a bus, for example), but there may be situations where latency causes problems. One possible case is when you have multiple channels containing some of the same sound source (a live drum recording for example, with a great amount of snare leakage into cymbal microphones). It can happen that the time shift introduces phasing effects when you place a latent plugin on one of these tracks.

The control output named "latency" is provided to solve this problem. The exact amount of latency is always written to this output during operation. Advanced hosts can shift the entire track backwards in time, based on this value. Or they may delay other tracks artificially so there is no time shift error between tracks. This compensation of LADSPA plugin latency works well in Ardour, so if you are an Ardour user, you have nothing to do with latency. Note that contrary to other control outputs, Ardour does not display the one named "latency" -- so you won't actually see the latency output.

If you are using another host, you may want to check if your host supports this kind of latency compensation. If not, it is recommended to shift the track backwards manually, by the number of samples indicated by the "latency" output (provided that your host displays this output, which is likely if it does not use it). Fortunately, latency is not an issue when using a plugin on master outputs or processing separate tracks outside the context of a mix.