Converting from audio signals to numeric control streams

We sometimes need to convert in the other direction, from an audio signal to a sporadic one. To go in this direction, we somehow provide a series of logical times (a time sequence), as well as an audio signal. For output we want a control stream combining the time sequence with values taken from the audio signal. We do this when we want to incorporate the signal's value as part of a control computation.

For example, we might be controlling the amplitude of a signal using a line~ object as in Example A03.line.pd (Page ). Suppose we wish to turn off the sound at a fixed rate of speed instead of in a fixed amount of time. For instance, we might want to re-use the network for another sound and wish to mute it as quickly as possible without audible artifacts; we probably can ramp it off in less time if the current amplitude is low than if it is high. To do this we must confect a message to the line~ object to send it to zero in an amount of time we'll calculate on the basis of its current output value. This will require, first of all, that we "sample" the line~ object's output (an audio signal) into a control stream.

The same issues of time delay and accuracy appear as for sporadic to audio conversion. Again there will be a tradeoff between immediacy and accuracy. Suppose as before that we are calculating audio in blocks of 4 samples, and suppose that at logical time 6 we want to look at the value of an audio signal, and use it to change the value of another one. As shown in Figure 3.2 (part b), the most recently calculated value of the signal will be for index 3 and the earliest index at which our calculation can affect a signal is 4. We can therefore carry out the whole affair with a delay of only one sample. However, we can't choose exactly which sample--the update can occur only at a block boundary.

As before, we can trade immediacy for increased time accuracy. If it matters exactly at which sample we carry out the audio-to-control-to-audio computation, we read the sample of index 2 and update the one at index 6. Then if we want to do the same thing again at logical time 7, we read from index 3 and update at index 7, and so on. In general, if the block size is $B$, and for any index $n$, we can always read the sample at index $n-B$ and affect the one at index $n$. There is thus a round-trip delay of $B$ samples in going from audio to control to audio computation, which is the price incurred for being able to name the index $n$ exactly.

If we wish to go further, to being able to specify a fraction of a sample, then (as before) we can use interpolation--at a slight further increase in delay. In general, as in the case of sporadic-to-audio conversion, in most cases the simplest solution is the best, but occasionally we have to do extra work.