Buffering interface

The Sound and Filterbank classes (and all classes derived from them) all implement the same buffering mechanism. The purpose of this is to allow for efficient processing of multiple channels in buffers. Rather than precomputing the application of filters to all channels (which for large numbers of channels or long sounds would not fit in memory), we process small chunks at a time. The entire design of these classes is based on the idea of buffering, as defined by the base class Bufferable (see section Class diagram). Each class has two methods, buffer_init() to initialise the buffer, and buffer_fetch(start, end) to fetch the portion of the buffer from samples with indices from start to end (not including end as standard for Python). The buffer_fetch(start, end) method should return a 2D array of shape (end-start, nchannels) with the buffered values.

From the user point of view, all you need to do, having set up a chain of Sound and Filterbank objects, is to call buffer_fetch(start, end) repeatedly. If the output of a Filterbank is being plugged into a FilterbankGroup object, everything is handled automatically. For cases where the number of channels is small or the length of the input source is short, you can use the Filterbank.process() method to automatically handle the initialisation and repeated application of buffer_fetch.

To extend Filterbank, it is often sufficient just to implement the buffer_apply(input) method. See the documentation for Filterbank for more details.